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+
+
+**ITU-T**
+
+TELECOMMUNICATION  
+STANDARDIZATION SECTOR  
+OF ITU
+
+**G.1011**
+
+(07/2016)
+
+SERIES G: TRANSMISSION SYSTEMS AND MEDIA,  
+DIGITAL SYSTEMS AND NETWORKS
+
+Multimedia Quality of Service and performance – Generic  
+and user-related aspects
+
+---
+
+**Reference guide to quality of experience  
+assessment methodologies**
+
+Recommendation ITU-T G.1011
+
+# ITU-T G-SERIES RECOMMENDATIONS **TRANSMISSION SYSTEMS AND MEDIA, DIGITAL SYSTEMS AND NETWORKS**
+
+|                                                                                                                                              |                      |
+|----------------------------------------------------------------------------------------------------------------------------------------------|----------------------|
+| INTERNATIONAL TELEPHONE CONNECTIONS AND CIRCUITS                                                                                             | G.100–G.199          |
+| GENERAL CHARACTERISTICS COMMON TO ALL ANALOGUE CARRIER-TRANSMISSION SYSTEMS                                                                  | G.200–G.299          |
+| INDIVIDUAL CHARACTERISTICS OF INTERNATIONAL CARRIER TELEPHONE SYSTEMS ON METALLIC LINES                                                      | G.300–G.399          |
+| GENERAL CHARACTERISTICS OF INTERNATIONAL CARRIER TELEPHONE SYSTEMS ON RADIO-RELAY OR SATELLITE LINKS AND INTERCONNECTION WITH METALLIC LINES | G.400–G.449          |
+| COORDINATION OF RADIOTELEPHONY AND LINE TELEPHONY                                                                                            | G.450–G.499          |
+| TRANSMISSION MEDIA AND OPTICAL SYSTEMS CHARACTERISTICS                                                                                       | G.600–G.699          |
+| DIGITAL TERMINAL EQUIPMENTS                                                                                                                  | G.700–G.799          |
+| DIGITAL NETWORKS                                                                                                                             | G.800–G.899          |
+| DIGITAL SECTIONS AND DIGITAL LINE SYSTEM                                                                                                     | G.900–G.999          |
+| <b>MULTIMEDIA QUALITY OF SERVICE AND PERFORMANCE – GENERIC AND USER-RELATED ASPECTS</b>                                                      | <b>G.1000–G.1999</b> |
+| TRANSMISSION MEDIA CHARACTERISTICS                                                                                                           | G.6000–G.6999        |
+| DATA OVER TRANSPORT – GENERIC ASPECTS                                                                                                        | G.7000–G.7999        |
+| PACKET OVER TRANSPORT ASPECTS                                                                                                                | G.8000–G.8999        |
+| ACCESS NETWORKS                                                                                                                              | G.9000–G.9999        |
+
+*For further details, please refer to the list of ITU-T Recommendations.*
+
+## Recommendation ITU-T G.1011
+
+# Reference guide to quality of experience assessment methodologies
+
+## Summary
+
+Recommendation ITU-T G.1011 provides a reference guide to existing standards for quality of experience (QoE) assessment methodologies. This Recommendation specifies QoE assessment approaches and classifications of different applications, and identifies a taxonomy of QoE assessment standards with different technical categorizations.
+
+## History
+
+| Edition | Recommendation | Approval   | Study Group | Unique ID*                                                                |
+|---------|----------------|------------|-------------|---------------------------------------------------------------------------|
+| 1.0     | ITU-T G.1011   | 2010-06-29 | 12          | <a href="http://handle.itu.int/11.1002/1000/10864">11.1002/1000/10864</a> |
+| 2.0     | ITU-T G.1011   | 2013-05-14 | 12          | <a href="http://handle.itu.int/11.1002/1000/11931">11.1002/1000/11931</a> |
+| 3.0     | ITU-T G.1011   | 2015-06-29 | 12          | <a href="http://handle.itu.int/11.1002/1000/12507">11.1002/1000/12507</a> |
+| 4.0     | ITU-T G.1011   | 2016-07-29 | 12          | <a href="http://handle.itu.int/11.1002/1000/12966">11.1002/1000/12966</a> |
+
+## Keywords
+
+QoE, Quality of experience.
+
+---
+
+\* To access the Recommendation, type the URL <http://handle.itu.int/> in the address field of your web browser, followed by the Recommendation's unique ID. For example, <http://handle.itu.int/11.1002/1000/11830-en>.
+
+## FOREWORD
+
+The International Telecommunication Union (ITU) is the United Nations specialized agency in the field of telecommunications, information and communication technologies (ICTs). The ITU Telecommunication Standardization Sector (ITU-T) is a permanent organ of ITU. ITU-T is responsible for studying technical, operating and tariff questions and issuing Recommendations on them with a view to standardizing telecommunications on a worldwide basis.
+
+The World Telecommunication Standardization Assembly (WTSA), which meets every four years, establishes the topics for study by the ITU-T study groups which, in turn, produce Recommendations on these topics.
+
+The approval of ITU-T Recommendations is covered by the procedure laid down in WTSA Resolution 1.
+
+In some areas of information technology which fall within ITU-T's purview, the necessary standards are prepared on a collaborative basis with ISO and IEC.
+
+## NOTE
+
+In this Recommendation, the expression "Administration" is used for conciseness to indicate both a telecommunication administration and a recognized operating agency.
+
+Compliance with this Recommendation is voluntary. However, the Recommendation may contain certain mandatory provisions (to ensure, e.g., interoperability or applicability) and compliance with the Recommendation is achieved when all of these mandatory provisions are met. The words "shall" or some other obligatory language such as "must" and the negative equivalents are used to express requirements. The use of such words does not suggest that compliance with the Recommendation is required of any party.
+
+## INTELLECTUAL PROPERTY RIGHTS
+
+ITU draws attention to the possibility that the practice or implementation of this Recommendation may involve the use of a claimed Intellectual Property Right. ITU takes no position concerning the evidence, validity or applicability of claimed Intellectual Property Rights, whether asserted by ITU members or others outside of the Recommendation development process.
+
+As of the date of approval of this Recommendation, ITU had not received notice of intellectual property, protected by patents, which may be required to implement this Recommendation. However, implementers are cautioned that this may not represent the latest information and are therefore strongly urged to consult the TSB patent database at <http://www.itu.int/ITU-T/ipr/>.
+
+© ITU 2016
+
+All rights reserved. No part of this publication may be reproduced, by any means whatsoever, without the prior written permission of ITU.
+
+## Table of Contents
+
+###### Page
+
+|      |                                                            |    |
+|------|------------------------------------------------------------|----|
+| 1    | Scope.....                                                 | 1  |
+| 2    | References.....                                            | 1  |
+| 3    | Definitions .....                                          | 4  |
+| 3.1  | Terms defined elsewhere .....                              | 4  |
+| 3.2  | Terms defined in this Recommendation.....                  | 4  |
+| 4    | Abbreviations and acronyms .....                           | 4  |
+| 5    | Conventions .....                                          | 5  |
+| 6    | Overview of QoE assessment methods.....                    | 5  |
+| 7    | Concept.....                                               | 5  |
+| 7.1  | General approach to quality of experience assessment ..... | 5  |
+| 7.2  | Content dependency of QoE assessment.....                  | 9  |
+| 8    | Target services .....                                      | 9  |
+| 8.1  | Audio .....                                                | 9  |
+| 8.2  | Video .....                                                | 9  |
+| 8.3  | Data.....                                                  | 10 |
+| 9    | Measurement approaches for different scenarios .....       | 10 |
+| 9.1  | Media-layer models .....                                   | 10 |
+| 9.2  | Packet-layer models.....                                   | 11 |
+| 9.3  | Bitstream-layer models.....                                | 11 |
+| 9.4  | Hybrid models .....                                        | 12 |
+| 9.5  | Planning models .....                                      | 12 |
+| 10   | Taxonomy.....                                              | 12 |
+|      | Appendix I.....                                            | 17 |
+|      | Appendix II .....                                          | 20 |
+| II.1 | Metrics related to TCP based streaming video.....          | 20 |
+| II.2 | Metrics related to UDP based streaming video .....         | 20 |
+|      | Bibliography.....                                          | 21 |
+
+
+
+## Recommendation ITU-T G.1011
+
+## Reference guide to quality of experience assessment methodologies
+
+# 1 Scope
+
+Quality of experience (QoE) is a widely used term in the telecommunication industry. However, there is no single document available that describes methods for assessing QoE. This Recommendation provides a reference guide to existing standardized methods in ITU for QoE assessment. The overall approaches for the different assessment methodologies typically used are described, and a taxonomy of QoE assessment standards is defined.
+
+For different applications, this Recommendation identifies suitable subjective assessment methodologies, and where applicable, objective assessment methods that can be used to estimate subjective opinion, in addition to giving guidance on their usage and limitations.
+
+The intended users of this Recommendation include telecommunication carriers, service providers and equipment manufacturers.
+
+# 2 References
+
+The following ITU-T Recommendations and other references contain provisions which, through reference in this text, constitute provisions of this Recommendation. At the time of publication, the editions indicated were valid. All Recommendations and other references are subject to revision; users of this Recommendation are therefore encouraged to investigate the possibility of applying the most recent edition of the Recommendations and other references listed below. A list of the currently valid ITU-T Recommendations is regularly published. The reference to a document within this Recommendation does not give it, as a stand-alone document, the status of a Recommendation.
+
+- [ITU-T G.107] Recommendation ITU-T G.107 (2015), *The E-model: a computational model for use in transmission planning*.
+- [ITU-T G.107.1] Recommendation ITU-T G.107.1 (2015), *Wideband E-model*.
+- [ITU-T G.1030] Recommendation ITU-T G.1030 (2014), *Estimating end-to-end performance in IP networks for data applications*.
+- [ITU-T G.1070] Recommendation ITU-T G.1070 (2012), *Opinion model for video-telephony applications*.
+- [ITU-T G.1071] Recommendation ITU-T G.1071 (2015), *Opinion model for network planning of video and audio streaming applications*.
+- [ITU-T J.140] Recommendation ITU-T J.140 (1998), *Subjective picture quality assessment for digital cable television systems*.
+- [ITU-T J.144] Recommendation ITU-T J.144 (2004), *Objective perceptual video quality measurement techniques for digital cable television in the presence of a full reference*.
+- [ITU-T J.246] Recommendation ITU-T J.246 (2008), *Perceptual visual quality measurement techniques for multimedia services over digital cable television networks in the presence of a reduced bandwidth reference*.
+- [ITU-T J.247] Recommendation ITU-T J.247 (2008), *Objective perceptual multimedia video quality measurement in the presence of a full reference*.
+- [ITU-T J.249] Recommendation ITU-T J.249 (2010), *Perceptual video quality measurement techniques for digital cable television in the presence of a reduced reference*.
+
+- [ITU-T J.341] Recommendation ITU-T J.341 (2016), *Objective perceptual multimedia video quality measurement of HDTV for digital cable television in the presence of a full reference.*
+- [ITU-T J.342] Recommendation ITU-T J.342 (2011), *Objective multimedia video quality measurement of HDTV for digital cable television in the presence of a reduced reference signal.*
+- [ITU-T J.343] Recommendation ITU-T J.343 (2014), *Hybrid perceptual bitstream models for objective video quality measurements.*
+- [ITU-T J.343.1] Recommendation ITU-T J.343.1 (2014), *Hybrid-NRe objective perceptual video quality measurement for HDTV and multimedia IP-based video services in the presence of encrypted bitstream data.*
+- [ITU-T J.343.2] Recommendation ITU-T J.343.2 (2014), *Hybrid-NR objective perceptual video quality measurement for HDTV and multimedia IP-based video services in the presence of non-encrypted bitstream data.*
+- [ITU-T J.343.3] Recommendation ITU-T J.343.3 (2014), *Hybrid-RRe objective perceptual video quality measurement for HDTV and multimedia IP-based video services in the presence of a reduced reference signal and encrypted bitstream data.*
+- [ITU-T J.343.4] Recommendation ITU-T J.343.4 (2014), *Hybrid-RR objective perceptual video quality measurement for HDTV and multimedia IP-based video services in the presence of a reduced reference signal and non-encrypted bitstream data.*
+- [ITU-T J.343.5] Recommendation ITU-T J.343.5 (2014), *Hybrid-FRe objective perceptual video quality measurement for HDTV and multimedia IP-based video services in the presence of a full reference signal and encrypted bitstream data.*
+- [ITU-T J.343.6] Recommendation ITU-T J.343.6 (2014), *Hybrid-FR objective perceptual video quality measurement for HDTV and multimedia IP-based video services in the presence of a full reference signal and non-encrypted bitstream data.*
+- [ITU-T P.561] Recommendation ITU-T P.561 (2002), *In-service non-intrusive measurement device – Voice service measurements.*
+- [ITU-T P.562] Recommendation ITU-T P.562 (2004), *Analysis and interpretation of INMD voice-service measurements.*
+- [ITU-T P.563] Recommendation ITU-T P.563 (2004), *Single-ended method for objective speech quality assessment in narrow-band telephony applications.*
+- [ITU-T P.564] Recommendation ITU-T P.564 (2007), *Conformance testing for voice over IP transmission quality assessment models.*
+- [ITU-T P.800] Recommendation ITU-T P.800 (1996), *Methods for subjective determination of transmission quality.*
+- [ITU-T P.800.1] Recommendation ITU-T P.800.1 (2016), *Mean opinion score (MOS) terminology.*
+- [ITU-T P.800.2] Recommendation ITU-T P.800.2 (2016), *Mean opinion score interpretation and reporting.*
+- [ITU-T P.805] Recommendation ITU-T P.805 (2007), *Subjective evaluation of conversational quality.*
+- [ITU-T P.806] Recommendation ITU-T P.806 (2014), *A subjective quality test methodology using multiple rating scales.*
+
+- [ITU-T P.830] Recommendation ITU-T P.830 (1996), *Subjective performance assessment of telephone-band and wideband digital codecs.*
+- [ITU-T P.835] Recommendation ITU-T P.835 (2003), *Subjective test methodology for evaluating speech communication systems that include noise suppression algorithm.*
+- [ITU-T P.862] Recommendation ITU-T P.862 (2001), *Perceptual evaluation of speech quality (PESQ): An objective method for end-to-end speech quality assessment of narrow-band telephone networks and speech codecs.*
+- [ITU-T P.862.1] Recommendation ITU-T P.862.1 (2003), *Mapping function for transforming P.862 raw result scores to MOS-LQO.*
+- [ITU-T P.862.2] Recommendation ITU-T P.862.2 (2007), *Wideband extension to Recommendation P.862 for the assessment of wideband telephone networks and speech codecs.*
+- [ITU-T P.863] Recommendation ITU-T P.863 (2014), *Perceptual objective listening quality assessment.*
+- [ITU-T P.910] Recommendation ITU-T P.910 (2008), *Subjective video quality assessment methods for multimedia applications.*
+- [ITU-T P.911] Recommendation ITU-T P.911 (1998), *Subjective audiovisual quality assessment methods for multimedia applications.*
+- [ITU-T P.920] Recommendation ITU-T P.920 (2000), *Interactive test methods for audiovisual communications.*
+- [ITU-T P.1201.1] Recommendation ITU-T P.1201.1 (2012), *Parametric non-intrusive assessment of audiovisual media streaming quality – Lower resolution application area.*
+- [ITU-T P.1201.2] Recommendation ITU-T P.1201.2 (2012), *Parametric non-intrusive assessment of audiovisual media streaming quality – Higher resolution application area.*
+- [ITU-T P.1202.1] Recommendation ITU-T P.1202.1 (2012), *Parametric non-intrusive bitstream assessment of video media streaming quality – Lower resolution application area.*
+- [ITU-T P.1202.2] Recommendation ITU-T P.1202.2 (2013), *Parametric non-intrusive bitstream assessment of video media streaming quality – Higher resolution application area.*
+- [ITU-T P.1301] Recommendation ITU-T P.1301 (2012), *Subjective quality evaluation of audio and audiovisual multiparty telemeetings.*
+- [ITU-T P.1302] Recommendation ITU-T P.1302 (2014), *Subjective method for simulated conversation tests addressing speech and audiovisual call quality.*
+- [ITU-T P.1311] Recommendation ITU-T P.1311 (2014), *Method for determining the intelligibility of multiple concurrent talkers.*
+- [ITU-T P.1312] Recommendation ITU-T P.1312 (2016), *Method for the measurement of the communication effectiveness of multiparty telemeetings using task performance.*
+- [ITU-T P.1501] Recommendation ITU-T P.1501 (2014), *Subjective testing methodology for web browsing.*
+
+- [ITU-R BS.1116-3] Recommendation ITU-R BS.1116-3 (2015), *Methods for the subjective assessment of small impairments in audio systems*.
+- [ITU-R BS.1285] Recommendation ITU-R BS.1285 (1997), *Pre-selection methods for the subjective assessment of small impairments in audio systems*.
+- [ITU-R BS.1387-1] Recommendation ITU-R BS.1387-1 (2001), *Method for objective measurements of perceived audio quality*.
+- [ITU-R BS.1534-3] Recommendation ITU-R BS.1534-3 (2015), *Method for the subjective assessment of intermediate quality level of audio systems*.
+- [ITU-R BT.500-13] Recommendation ITU-R BT.500-13 (2012), *Methodology for the subjective assessment of the quality of television pictures*.
+
+# 3 Definitions
+
+## 3.1 Terms defined elsewhere
+
+This Recommendation uses the following term defined elsewhere:
+
+**3.1.1 quality of experience** [b-ITU-T P.10 Amd.2]: The overall acceptability of an application or service, as perceived subjectively by the end-user.
+
+NOTE 1 – Quality of experience includes the complete end-to-end system effects (client, terminal, network, services infrastructure, etc.).
+
+NOTE 2 – Overall acceptability may be influenced by user expectations and context.
+
+## 3.2 Terms defined in this Recommendation
+
+None.
+
+# 4 Abbreviations and acronyms
+
+This Recommendation uses the following abbreviations and acronyms:
+
+|                       |                                                   |
+|-----------------------|---------------------------------------------------|
+| CIF                   | Common Intermediate Format                        |
+| FR                    | Full Reference                                    |
+| HDTV                  | High Definition Television                        |
+| HVGA                  | Half Video Graphics Array                         |
+| IPTV                  | Internet Protocol Television                      |
+| ISDN                  | Integrated Services Digital Network               |
+| MOS                   | Mean Opinion Score                                |
+| MOS <sub>A,V,MM</sub> | Mean Opinion Score for Audio, Video or Multimedia |
+| NB                    | Narrowband                                        |
+| NR                    | No Reference                                      |
+| PSTN                  | Public Switched Telephone Network                 |
+| QCIF                  | Quarter Common Intermediate Format                |
+| QoE                   | Quality of Experience                             |
+| QVGA                  | Quarter Video Graphics Array                      |
+| RR                    | Reduced Reference                                 |
+
+|      |                              |
+|------|------------------------------|
+| SD   | Standard Definition          |
+| SWB  | Super Wideband               |
+| VGA  | Video Graphics Array         |
+| VoIP | Voice over Internet Protocol |
+| WB   | Wideband                     |
+
+# 5 Conventions
+
+This Recommendation uses the following conventions:
+
+LAB laboratory testing;
+
+MON monitoring;
+
+PLN planning.
+
+# 6 Overview of QoE assessment methods
+
+This Recommendation gives a general overview of QoE assessment methods. For example, the concept of QoE assessment; different approaches to assessment, whether assessment is subjective or objective; target services, e.g., streaming video or web browsing; measurement approaches for different scenarios; and definitions of metrics for assessment models. Figure 6-1 shows the structure of the content in this Recommendation.
+
+![Figure 6-1: Structure of the contents of this Recommendation. The diagram is titled 'ITU-T G.1011 structure' and shows a hierarchical breakdown of the recommendation's content. It starts with 'Concept' at the top, branching into 'Definition' (referencing ITU-T P.800.1, P.800.2) and 'General approach' (Subjective e.g., ITU-T P.911 and Objective). The 'Objective' approach further branches into 'Monitoring' (e.g., ITU-T P.1202.1), 'Planning' (e.g., ITU-T G.1071), and 'Lab-testing' (e.g., ITU-T J.341). A 'Measurement approach' branches into 'Media-layer model', 'Bitstream-layer model', and 'Planning model'. 'Media-layer model' further branches into 'Packet-layer model' and 'Hybrid model'. Below this, 'Existing QoE assessment standards' are categorized into 'Audio', 'Video', 'Multimedia', and 'Data', each with their respective subjective and objective standards (e.g., ITU-T P.830, BS. 1116-3, etc. for Audio; ITU-T P.910, ITU-T J.140, etc. for Video; ITU-T P.911, etc. for Multimedia; ITU-T P.1501, etc. for Data).](c78c2eefd86269d1740ab85a916f24f2_img.jpg)
+
+```
+
+graph TD
+    Concept[Concept] --> Definition[Definition]
+    Concept --> General_approach[General approach]
+    Definition --> Def_Examples["e.g., ITU-T P.800.1,  
+ITU-T P.800.2"]
+    General_approach --> Subjective["Subjective  
+e.g., ITU-T P.911"]
+    General_approach --> Objective[Objective]
+    Objective --> Monitoring["Monitoring  
+e.g., ITU-T P.1202.1"]
+    Objective --> Planning["Planning  
+e.g., ITU-T G.1071"]
+    Objective --> Lab_testing["Lab-testing  
+e.g., ITU-T J.341"]
+    Media_layer_model[Media-layer model] --> Packet_layer_model[Packet-layer model]
+    Media_layer_model --> Hybrid_model[Hybrid model]
+    Bitstream_layer_model[Bitstream-layer model]
+    Planning_model[Planning model]
+    Measurement_approach[Measurement approach] --> Media_layer_model
+    Measurement_approach --> Bitstream_layer_model
+    Measurement_approach --> Planning_model
+    Existing_standards[Existing QoE assessment standards] --> Audio[Audio]
+    Existing_standards --> Video[Video]
+    Existing_standards --> Multimedia[Multimedia]
+    Existing_standards --> Data[Data]
+    Audio --> Audio_Subjective["Subjective  
+ITU-T P.830,  
+BS. 1116-3, etc.  
+Objective  
+BS.1387-1, etc."]
+    Video --> Video_Subjective["Subjective  
+ITU-T P.910,  
+ITU-T J.140, etc.  
+Objective  
+ITU-T J.144,  
+ITU-T J.247, etc."]
+    Multimedia --> Multimedia_Subjective["Subjective  
+ITU-T P.911, etc.  
+Objective  
+ITU-T P.1201.1,  
+ITU-T P.1201.2, etc."]
+    Data --> Data_Subjective["Subjective  
+ITU-T P.1501, etc.  
+Objective  
+ITU-T G.1030, etc."]
+  
+```
+
+Figure 6-1: Structure of the contents of this Recommendation. The diagram is titled 'ITU-T G.1011 structure' and shows a hierarchical breakdown of the recommendation's content. It starts with 'Concept' at the top, branching into 'Definition' (referencing ITU-T P.800.1, P.800.2) and 'General approach' (Subjective e.g., ITU-T P.911 and Objective). The 'Objective' approach further branches into 'Monitoring' (e.g., ITU-T P.1202.1), 'Planning' (e.g., ITU-T G.1071), and 'Lab-testing' (e.g., ITU-T J.341). A 'Measurement approach' branches into 'Media-layer model', 'Bitstream-layer model', and 'Planning model'. 'Media-layer model' further branches into 'Packet-layer model' and 'Hybrid model'. Below this, 'Existing QoE assessment standards' are categorized into 'Audio', 'Video', 'Multimedia', and 'Data', each with their respective subjective and objective standards (e.g., ITU-T P.830, BS. 1116-3, etc. for Audio; ITU-T P.910, ITU-T J.140, etc. for Video; ITU-T P.911, etc. for Multimedia; ITU-T P.1501, etc. for Data).
+
+Figure 6-1 – Structure of the contents of this Recommendation
+
+# 7 Concept
+
+## 7.1 General approach to quality of experience assessment
+
+In principle, assessment of quality of experience (QoE) must be performed using subjective tests with metrics, such as the mean opinion score (MOS). However, it is also possible and sometimes more
+
+convenient to estimate QoE based on objective testing and associated quality estimation models. Through different quality estimation models, it is possible to measure or calculate the objective parameters affecting QoE, in order to evaluate QoE. Subjective testing needs more resources and effort, because it requires human subjects. On the other hand, objective measurement and automatic calculation using appropriate quality estimation models is generally much faster and cheaper, but the accuracy of the final evaluation depends on the accuracy of these models.
+
+In general, if QoE is evaluated objectively, it may be tested in intrusive mode, non-intrusive mode or planning mode. "Intrusive mode" means that the quality assessment system requires that a signal be injected into the system under test in order to generate a degraded output signal. This implies that the channel must be taken out of service for normal traffic. Conversely, for "non-intrusive mode", the quality assessment system can be used while live traffic is carried by the channel, without the need for any active test signals. "Planning mode" is not used in a real-time environment, but as a tool for the design of systems, and hence does not require any real-time inputs.
+
+Irrespective of which mode it is in, the most important thing is to identify the key objective parameters that affect QoE and then relate these parameters to QoE measurement indices through specific quality estimation models.
+
+Since QoE, i.e., perceived quality, is internal to the user, perceptual quality tests are ultimately the only means of validly and reliably assessing quality. Figure 7-1 gives a graphical explanation of the terms "valid" and "reliable".
+
+![Figure 7-1: Validity and reliability. The figure consists of four circular targets arranged in a 2x2 grid, each with concentric circles representing a bullseye. Blue 'X' marks represent data points. Top-left: 'high reliability, high validity' - points are tightly clustered in the center. Top-right: 'high reliability, low validity' - points are tightly clustered on the right side, away from the center. Bottom-left: 'low reliability, high validity' - points are scattered around the center. Bottom-right: 'low reliability, low validity' - points are scattered and mostly outside the target area. A small label 'G.1011(16)_F7-1' is in the bottom right corner.](252ea48d02dce93965b91746fb376f35_img.jpg)
+
+Figure 7-1: Validity and reliability. The figure consists of four circular targets arranged in a 2x2 grid, each with concentric circles representing a bullseye. Blue 'X' marks represent data points. Top-left: 'high reliability, high validity' - points are tightly clustered in the center. Top-right: 'high reliability, low validity' - points are tightly clustered on the right side, away from the center. Bottom-left: 'low reliability, high validity' - points are scattered around the center. Bottom-right: 'low reliability, low validity' - points are scattered and mostly outside the target area. A small label 'G.1011(16)\_F7-1' is in the bottom right corner.
+
+**Figure 7-1 – Validity and reliability**
+
+NOTE – Validity describes how well a method actually measures what it is intended to measure. Reliability describes the accuracy of a method in terms of the scattering of results, e.g., when test assessment is repeated ([b-Guilford, 1954], pp. 349–357). Nowadays, validity is often replaced with the term accuracy, and reliability is replaced with the term precision.
+
+The intrinsic variation that subjective test methods may be related with (e.g., from subject to subject or test laboratory to test laboratory) can be reduced based on a proper method design, yielding reliable and valid test results. There are a number of ITU Recommendations on how to properly design subjective tests, as summarized in clause 10.
+
+Although objective assessment methods will lead to a certain level of reliability (i.e., the same quality estimate for a given input whenever it is applied to that input), their validity and reliability may be limited. Such model limitations are typically reflected in the corresponding Recommendations (see clause 10 for a summary of ITU Recommendations related to objective models). Within the range of system configurations, the objective methods that have been designed for, and proven to provide, valid and reliable quality predictions, can be used as a replacement for or a complement to subjective tests.
+
+All methods can be classified according to how well diagnostic information can be obtained:
+
+- Glass-box approaches enable insight into the internal functionality of the system. Such insight, in turn, may give access to diagnostic information. Hence, the source of a certain quality level may be identified and the relationship between system configuration and quality can be observed. For example, parameter-based models are typically glass-box models, since they build on a "diagnostic" description of the system.
+- Black-box approaches do not enable insight into system internals. Hence, with black-box approaches direct indication of the source of low quality is not available. Certain signal-based models are typical examples. For black-box methods, the system under test is described only by its input and output. Based on additional processing of the results, diagnostic information may be inferred.
+
+Current objective quality assessment techniques cannot necessarily be classified in a strict manner according to this black-box/glass-box dichotomy. Instead, different methods enable more or less diagnostic information to be obtained. Sometimes the terms "white box" and "grey box" are used in the literature in these cases ([b-Möller, 2005], pp. 86-89).
+
+### 7.1.1 Subjective assessment
+
+According to [b-Jekosch, 2005], p. 91, a (speech) quality test is defined as:
+
+"[a] routine procedure for examining one or more empirically restrictive quality features of perceived speech with the aim of making a quantitative statement on these features".
+
+Using this definition, subjective test methods can be broadly categorized as utilitarian and analytic test methods on the basis of whether the totality or a certain subset of perceived quality features is to be investigated (analytical), or whether single individual features or integral quality are to be measured (utilitarian).
+
+Utilitarian test methods employ a unidimensional quality-rating scale, so that a direct comparison can be made between the qualities of different systems. In turn, analytical methods are related to a multidimensional analysis of test stimuli: They aim at identifying and/or quantifying some or all perceptual features underlying quality.
+
+In order to capture the entire range of underlying auditory features, and to ensure a common understanding of these features, the reliability and validity of analytical methods can be significantly improved when trained listeners are used (e.g., [b-Mattila, 2001]; [b-Möller, 2000], pp. 105-120). In turn, tests of the utilitarian type are typically carried out with naive subjects. These should ideally represent the user group at which the system or service under investigation is aimed (age range, social and cultural background, etc.).
+
+It should be noted that this utilitarian-analytical categorization of auditory test methods can be considered as being of utilitarian type itself, as it employs a unidimensional view on test methods. A multidimensional, that is, analytical view on speech quality assessment methods was provided by [b-Jekosch, 2005] (pp. 105-111).
+
+Utilitarian methods have three main goals: "To be reasonably efficient in test administration and data analysis, to measure speech quality on a unidimensional scale, and to have good reliability in the test method" ([b-Quackenbush, 1988], pp. 15-16).
+
+All test methods can be classified according to the categorization generally applicable to psychometric methods (after [b-Möller, 2000], pp. 48-49):
+
+- The applied scaling method and scale level, with the scale levels being (after [b-Stevens, 1951], p. 25):
+  - 1) Nominal (identifying equality);
+  - 2) Ordinal (identifying greater or less);
+
+- 3) Interval (identifying equality of intervals or differences);
+  - 4) Ratio (identifying equality of ratios).
+- The presentation method used, such as:
+    - Adjustment, i.e., the stimulus is changed to meet a certain description or judgement;
+    - Constancy, i.e., one or several stimuli are presented per trial and a description or judgement is given (single stimulus, paired comparison, etc.).
+  - The test "modality":
+    - Passive, e.g., listening or viewing;
+    - Active, e.g., talking or moving;
+    - Interactive, e.g., in a conversation.
+  - The instance at which quality is being judged:
+    - Judgement after stimulus/stimuli presentation;
+    - Continuous judgement during stimulus/stimuli presentation (e.g., in case of time-varying degradations).
+
+### 7.1.2 Objective assessment
+
+In order to reduce the necessity for time-consuming and costly perception tests to measure the quality of networks and systems, much effort has been spent on the development of alternative, instrumental, i.e., "objective" methods. It has to be noted, however, that the speech quality of today's telecommunication systems cannot readily be determined solely on the basis of basic signal measures like the signal-to-noise ratio. Although quality may be correlated with different instrumentally measurable signal characteristics, it is typically not possible to establish a simple relationship between these instrumentally measurable magnitudes and the quality perceived by a user of the system.
+
+To account for the more complex interdependencies, different types of quality estimation and prediction models have been developed. Each model has its proper domain of application, and range of system or service conditions it has been designed for. Consequently, there is no universal quality model that can be applied in all circumstances. In order to better determine which model to choose for a given application, the models can be categorized according to different criteria (adapted from [b-Möller, 2002]):
+
+- what the application is aimed at (network planning, codec optimization, service monitoring, etc.);
+- the predicted quality features (integral quality – audio, video, audiovisual, speech, etc.), intelligibility, conversational aspects, etc.;
+- the network components and configuration under consideration (entire connection end-to-end, codecs, etc.);
+- the respective model input parameters (measured parameters like noise levels, information extracted from protocol headers like loss rates, entire signals, etc.);
+- the way in which the input information is obtained:
+  - by using passive measurements, which do not interfere with the running system or service, but involve an observation of the system or service during its normal operation;
+  - by using active measurements, which involve dedicated measurement signals or data being processed by the system, and evaluating the transmitted result;
+  - by estimation, e.g., during the network or service planning phase.
+- the level of interactivity of the service aspects that can be assessed with the model, related to:
+  - a passive perception (listening, viewing, etc.);
+
+- an active usage (talking, moving);
+  - an interactive usage (e.g., for a conversation).
+- the extent to which psychophysical knowledge or rather purely empirical data have been incorporated.
+
+## 7.2 Content dependency of QoE assessment
+
+QoE evaluation depends seriously on the context of subjective/objective experiments. [ITU-T P.800.1] provides various indices for MOS that clarify the context behind the MOS, which is one of the most widely used metrics. This metric is derived from the five-point ACR scale defined by several ITU-T P.800 series Recommendations (e.g., [ITU-T P.800]), ITU-T P.900 series Recommendations (e.g., [ITU-T P.910]) and [ITU-R BT.500-13]. For example, MOS-LQSW means that MOS was obtained in a "listening-only (one-way) test" (by letters "LQ"), and that the test was conducted in a subjective manner (by letter "S"), and that it was obtained for wideband (50-7000 Hz) speech relative to a wideband high quality reference.
+
+[ITU-T P.800.2] gives additional details on how to provide contextual information regarding the reported MOS. For example, [ITU-T P.800.2] recommends reporting acoustic/viewing environment, audio/video material, and listening/viewing devices used in the experiment.
+
+When one interprets the reported MOS, he or she needs to be careful about the context of an experiment, in which the MOS was derived.
+
+# 8 Target services
+
+This Recommendation gives guidelines for QoE assessment of various telecommunication services mainly utilizing audio and visual media.
+
+## 8.1 Audio
+
+- Conversational voice and voice messaging
+  - Speech communication services, such as mobile telephony and voice over Internet protocol (VoIP), as well as conventional public switched telephone network (PSTN) services, are important targets of this Recommendation. The speech bandwidth can be either narrowband (NB) (300-3400 Hz) or wideband (WB) (100-7000 Hz).
+- Streaming/on-demand audio
+  - Audio delivery services provide audio content through telecommunication networks. They can be either download based or streaming based. Content is usually, but not limited to, musical material. Audio bandwidth can be from NB to full band (20-20000 Hz), although usually either super wideband (SWB) (50-14000 Hz) or full band (20-20000 Hz) is used.
+
+## 8.2 Video
+
+- Videophone/videoconferencing
+  - Audiovisual communication refers to two-way interactive communication using speech and video signals, such as video telephony and videoconferencing. The bandwidth of speech signals can be either NB (300-3400 Hz) or WB (100-7000 Hz). Video resolution can range between quarter common intermediate format (QCIF) and high definition television (HDTV).
+- Streaming/on-demand video
+  - Video delivery services provide video content (normally with audio signals) through telecommunication networks. They can be either download based or streaming based. Typical content includes movies, documentaries, sports, news, and advertisement. Audio
+
+bandwidth can be from NB (300-3400 Hz) to full band (20-20000 Hz), although usually either SWB (50-14000 Hz) or full band (20-20000 Hz) is used. Video resolution can be between QCIF and HDTV.
+
+## 8.3 Data
+
+- Data communication services targeted in this Recommendation are primarily web browsing.
+
+# 9 Measurement approaches for different scenarios
+
+Basically, there are five types of objective quality assessment methodologies. Depending on the application, these can be divided into media-layer models, packet-layer models, bitstream models, hybrid models, and planning models.
+
+Objective quality assessment models can be used for several purposes:
+
+- Planning: "Planning" refers to estimating the perceived quality of services of networks/systems before they are implemented. Since it is not used in a real-time environment, no real-time inputs are required to the objective model.
+- Lab-testing: "Lab-testing" refers to estimating the perceived quality of services of networks/systems in the laboratory while the equipment is being developed.
+- Monitoring: "Monitoring" refers to estimating the perceived quality of services of networks/systems that are operational. Necessary information is collected from the network and analysed to reflect the degradation of the quality experienced by users.
+
+## 9.1 Media-layer models
+
+Media-layer models (Figure 9-1) use the actual media signals (audio/video) as their input, and may take into account codec compression and channel characteristics. They use complex perceptually based psychophysical models to estimate QoE, by comparing (full reference/reduced reference (FR/RR)) the output (degraded) signal to the input (clean) signal or just analysing the output (degraded) signal (no reference (NR)).
+
+The main applications of full reference models are QoE assessment in the laboratory, for example, codec comparison/optimization, since it uses not only the degraded (received) signal, but also the original (transmitted) signal to estimate QoE. Conversely, RR/NR models can be applied to QoE monitoring at the mid-point or end-points of the Internet protocol television (IPTV) network.
+
+![Figure 9-1: Media-layer models. This figure contains three separate flowcharts illustrating different media-layer modeling approaches. The top flowchart shows a 'Media signal' entering a 'System under test' to produce a 'Degraded signal'. Both signals are then fed into a 'Media-layer model (FR)', which outputs the 'Media quality'. The middle flowchart is similar but includes 'Side info.' as an additional input to the 'Media-layer model (RR)'. The bottom flowchart shows only the 'Degraded signal' being input to a 'Media-layer model (NR)', which also outputs the 'Media quality'. A label 'G.1011(16)_F9-1' is present at the bottom right of the diagrams.](8fbdfc3d17fb1dae7b2d8f5a287fa9fc_img.jpg)
+
+Figure 9-1: Media-layer models. This figure contains three separate flowcharts illustrating different media-layer modeling approaches. The top flowchart shows a 'Media signal' entering a 'System under test' to produce a 'Degraded signal'. Both signals are then fed into a 'Media-layer model (FR)', which outputs the 'Media quality'. The middle flowchart is similar but includes 'Side info.' as an additional input to the 'Media-layer model (RR)'. The bottom flowchart shows only the 'Degraded signal' being input to a 'Media-layer model (NR)', which also outputs the 'Media quality'. A label 'G.1011(16)\_F9-1' is present at the bottom right of the diagrams.
+
+**Figure 9-1 – Media-layer models**
+
+## 9.2 Packet-layer models
+
+Packet-layer models (Figure 9-2) utilize only packet header information for QoE prediction. Because they do not parse the packet payload information, it is difficult to incorporate in such models aspects of QoE that are related to the media content, although they have a very light measurement of computational efficiency. These models are primarily utilized as network probes at the mid-point or end-points of the network.
+
+![Figure 9-2: Packet-layer model. This flowchart shows a 'Parameter extraction function (packet header information)' block providing 'Input' to a 'Packet-layer model' block. The 'Packet-layer model' block also receives an 'Assumption' from below and produces an 'Output' labeled 'MOS_A, V, MM'. A label 'G.1011(16)_F9-2' is at the bottom right.](df82d77a0d2637cbf2da9ea920a554fa_img.jpg)
+
+Figure 9-2: Packet-layer model. This flowchart shows a 'Parameter extraction function (packet header information)' block providing 'Input' to a 'Packet-layer model' block. The 'Packet-layer model' block also receives an 'Assumption' from below and produces an 'Output' labeled 'MOS\_A, V, MM'. A label 'G.1011(16)\_F9-2' is at the bottom right.
+
+**Figure 9-2 – Packet-layer model**
+
+## 9.3 Bitstream-layer models
+
+A bitstream model (Figure 9-3) takes not only the encoded bitstream information, but also the packet header information as its input. Therefore, such a model can be viewed as a combination model between media-layer models and packet-layer models.
+
+![Figure 9-3: Bitstream-layer model. This flowchart shows a 'Packet (header and payload) information' block providing 'Input' to a 'Bitstream layer model' block. The 'Bitstream layer model' block produces an 'Output' labeled 'MOS_A, V, MM'. A label 'G.1011(16)_F9-3' is at the bottom right.](43837b056625d3d6ce615e4c02f163bb_img.jpg)
+
+Figure 9-3: Bitstream-layer model. This flowchart shows a 'Packet (header and payload) information' block providing 'Input' to a 'Bitstream layer model' block. The 'Bitstream layer model' block produces an 'Output' labeled 'MOS\_A, V, MM'. A label 'G.1011(16)\_F9-3' is at the bottom right.
+
+**Figure 9-3 – Bitstream-layer model**
+
+Since the bitstream-layer model uses only the received packet information (degraded signal), it can be applied to QoE monitoring at the mid-point or end-points of the IPTV network.
+
+## 9.4 Hybrid models
+
+A hybrid model (Figure 9-4), as its name implies, is the combination of the previously mentioned models. It employs as much information as possible to predict QoE. An example of these models is shown below:
+
+![Figure 9-4: Example of a hybrid model. A block diagram showing 'Packet information and media signals' as input to a 'Hybrid model' block, which outputs 'MOS_A, V, MM'. The diagram is labeled G.1011(16)_F9-4.](4356776ca004ecba5d599667a155d7d4_img.jpg)
+
+```
+graph LR; A[Packet information and media signals] -- Input --> B[Hybrid model]; B -- Output --> C["MOSA, V, MM"]; style C text-align: center; C --- D[G.1011(16)_F9-4]
+```
+
+Figure 9-4: Example of a hybrid model. A block diagram showing 'Packet information and media signals' as input to a 'Hybrid model' block, which outputs 'MOS\_A, V, MM'. The diagram is labeled G.1011(16)\_F9-4.
+
+Figure 9-4 – Example of a hybrid model
+
+## 9.5 Planning models
+
+The input for planning models (Figure 9-5) includes the quality planning parameters of networks or terminals. It usually requires prior knowledge about the system under test. Such models can be applied to network planning and terminal/application design.
+
+![Figure 9-5: Planning model. A block diagram showing 'Quality design parameters' as input to a 'Planning model' block, which also receives 'Planning assumptions' as input. The output is 'MOS_A, V, MM'. The diagram is labeled G.1011(16)_F9-5.](77464a47f104d0d647b2414591137b64_img.jpg)
+
+```
+graph LR; A[Quality design parameters] -- Input --> B[Planning model]; C[Planning assumptions] --> B; B -- Output --> D["MOSA, V, MM"]; style D text-align: center; D --- E[G.1011(16)_F9-5]
+```
+
+Figure 9-5: Planning model. A block diagram showing 'Quality design parameters' as input to a 'Planning model' block, which also receives 'Planning assumptions' as input. The output is 'MOS\_A, V, MM'. The diagram is labeled G.1011(16)\_F9-5.
+
+Figure 9-5 – Planning model
+
+Standard examples of such models are [ITU-T G.107] for speech and [ITU-T G.1070] for videophone.
+
+# 10 Taxonomy
+
+Existing QoE assessment standards are categorized in Table 10-1. Appendix I provides information on planned standards.
+
+**Table 10-1 – Current ITU Recommendations for QoE assessment**
+
+| Application                      | Media             | Conversational (CONV)/Non-conversational (NONCONV) | Subjective test methodology                                                                                            | Objective test methodology                                                                                                                         |                    |          |               |
+|----------------------------------|-------------------|----------------------------------------------------|------------------------------------------------------------------------------------------------------------------------|----------------------------------------------------------------------------------------------------------------------------------------------------|--------------------|----------|---------------|
+|                                  |                   |                                                    |                                                                                                                        | Model                                                                                                                                              | Transport protocol | FR/RR/NR | Primary usage |
+| Telephony                        | Speech            | NONCONV                                            | [ITU-T P.800]<br>[ITU-T P.830]<br>[ITU-T P.835]<br>[ITU-T P.1301]<br>[ITU-T P.1302]<br>[ITU-T P.1311]<br>[ITU-T P.806] | [ITU-T P.862] + [ITU-T P.862.1] (NB)<br>[ITU-T P.862.2] (WB)<br>[ITU-T P.863] (NB/WB/SWB)                                                          |                    | FR       | LAB, MON      |
+|                                  |                   |                                                    |                                                                                                                        | [ITU-T P.563] (NB)<br>[ITU-T P.564] (NB/WB)                                                                                                        |                    | NR       | MON           |
+|                                  |                   | CONV                                               | [ITU-T P.800]<br>[ITU-T P.805]<br>[ITU-T P.1301]<br>[ITU-T P.1312]                                                     | [ITU-T G.107] (NB)<br><b>[ITU-T G.107.1] (WB)</b>                                                                                                  |                    | NR       | PLN           |
+|                                  |                   |                                                    |                                                                                                                        | [ITU-T P.561] + [ITU-T P.562] (NB/WB)                                                                                                              |                    | NR       | MON           |
+| Video telephony                  | Multimedia (Note) | NONCONV                                            | <b>[ITU-T P.1302]</b>                                                                                                  |                                                                                                                                                    |                    |          |               |
+|                                  |                   | CONV                                               | [ITU-T P.920]<br>[ITU-T P.1301]                                                                                        | [ITU-T G.1070] (NB/WB)                                                                                                                             |                    | NR       | PLN           |
+| Video streaming (Mobile TV/IPTV) | Video             | NONCONV                                            | [ITU-T P.910]<br>[ITU-T J.140]<br>[ITU-R BT.500-13]                                                                    | [ITU-T J.144] standard definition (SD)<br>[ITU-T J.247] (QCIF, common intermediate format (CIF), video graphics array (VGA))<br>[ITU-T J.341] (HD) | Transport agnostic | FR       | LAB, MON      |
+
+**Table 10-1 – Current ITU Recommendations for QoE assessment**
+
+| Application                                                                                                                                  | Media      | Conversational (CONV)/Non-conversational (NONCONV) | Subjective test methodology                                                | Objective test methodology                                                                                                                                                                      |                    |          |               |
+|----------------------------------------------------------------------------------------------------------------------------------------------|------------|----------------------------------------------------|----------------------------------------------------------------------------|-------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------|--------------------|----------|---------------|
+|                                                                                                                                              |            |                                                    |                                                                            | Model                                                                                                                                                                                           | Transport protocol | FR/RR/NR | Primary usage |
+| Video streaming (Mobile TV/IPTV)                                                                                                             | Video      | NONCONV                                            | [ITU-T P.910]<br>[ITU-T J.140]<br>[ITU-R BT.500-13]                        | [ITU-T J.343.5] (HD, encrypted bitstream)<br>[ITU-T J.343.6] (HD, non-encrypted bitstream)                                                                                                      | UDP                | FR       | LAB, MON      |
+|                                                                                                                                              |            |                                                    |                                                                            | [ITU-T J.249] (SD)<br>[ITU-T J.246] (QCIF, CIF, VGA)<br>[ITU-T J.342] (HD)                                                                                                                      | Transport agnostic | RR       | MON           |
+|                                                                                                                                              |            |                                                    |                                                                            | [ITU-T J.343.3] (HD, encrypted bitstream)<br>[ITU-T J.343.4] (HD, non-encrypted bitstream)                                                                                                      | UDP                | RR       | MON           |
+|                                                                                                                                              |            |                                                    |                                                                            | [ITU-T J.343.1] (HD, encrypted bitstream)<br>[ITU-T J.343.2] (HD, non-encrypted bitstream)                                                                                                      | UDP                | NR       | MON           |
+|                                                                                                                                              | Audio      | NONCONV                                            | [ITU-T P.830]<br>[ITU-R BS.1116-3]<br>[ITU-R BS.1285]<br>[ITU-R BS.1534-3] | [ITU-R BS.1387-1]                                                                                                                                                                               | Transport agnostic | FR/RR    | MON/PLN       |
+|                                                                                                                                              |            |                                                    |                                                                            |                                                                                                                                                                                                 |                    |          |               |
+|                                                                                                                                              | Multimedia | NONCONV                                            | [ITU-T P.911]                                                              | [ITU-T P.1201.1] (QCIF, quarter video graphics array (QVGA), half video graphics array (HVGA))<br>[ITU-T P.1201.2] (SD, HD)<br>[ITU-T P.1202.1] (QCIF, QVGA, HVGA)<br>[ITU-T P.1202.2] (SD, HD) | UDP                | NR       | MON           |
+|                                                                                                                                              |            |                                                    |                                                                            | [ITU-T P.1201 Amd. 2], App. III                                                                                                                                                                 | TCP                | NR       | MON           |
+|                                                                                                                                              |            |                                                    |                                                                            | [ITU-T G.1071] (SD, HD)                                                                                                                                                                         | UDP                | NR       | PLN           |
+| Web browsing                                                                                                                                 | Data       |                                                    | [ITU-T P.1501]                                                             | [ITU-T G.1030]                                                                                                                                                                                  | TCP                | NR       | PLN           |
+| NOTE – For individual media (i.e., speech and video), the Recommendations used in telephony and video-streaming applications are applicable. |            |                                                    |                                                                            |                                                                                                                                                                                                 |                    |          |               |
+
+Table 10-2 gives detailed information on planning models, including existing and ongoing models.
+
+**Table 10-2 – Summary of existing planning models for video streaming applications**
+
+| Transport protocol | Model                               | Video/Audiovisual | Video resolution                                                                | Video codec                           | Video type |
+|--------------------|-------------------------------------|-------------------|---------------------------------------------------------------------------------|---------------------------------------|------------|
+| UDP                | ITU-T G.1071<br>(Lower resolution)  | Video/Audiovisual | HVGA, QVGA,<br>QCIF                                                             | MPEG-4 Part 2, H.264 (MPEG-4 Part 10) |            |
+|                    | ITU-T G.1071<br>(Higher resolution) | Video/Audiovisual | SD (PAL/NTSC),<br>HD (720p50, 720p60,<br>1080i50, 1080p25, 1080i60,<br>1080p30) | H.264 (MPEG-4 Part 10)                |            |
+| TCP                |                                     |                   |                                                                                 |                                       |            |
+
+Table 10-3 gives detailed information on monitoring models, including existing and ongoing models.
+
+**Table 10-3 – Summary of existing monitoring models for video streaming applications**
+
+| Transport protocol | Model          | Video/Audiovisual | Video resolution                                                                | Video codec                                                                                | Video type |
+|--------------------|----------------|-------------------|---------------------------------------------------------------------------------|--------------------------------------------------------------------------------------------|------------|
+| UDP                | ITU-T P.1201.1 | Audiovisual       | HVGA, QVGA,<br>QCIF                                                             | MPEG-4 Part 2, H.264 (MPEG-4 Part 10)                                                      | IPTV       |
+|                    | ITU-T P.1201.2 | Audiovisual       | SD (PAL/NTSC),<br>HD (720p50, 720p60,<br>1080i50, 1080p25, 1080i60,<br>1080p30) | H.264 (MPEG-4 Part 10)                                                                     | IPTV       |
+|                    | ITU-T J.343    | Video             | HD, VGA/WVGA                                                                    | H.264/AVC (MPEG-4 Part 10)                                                                 | IPTV       |
+|                    | ITU-T J.247    | Video             | QCIF, CIF, VGA                                                                  | H.264/AVC (MPEG-4 part 10), VC-1,<br>Windows Media 9, Real Video (RV 10),<br>MPEG-4 Part 2 |            |
+
+**Table 10-3 – Summary of existing monitoring models for video streaming applications**
+
+| <b>Transport protocol</b> | <b>Model</b>                  | <b>Video/Audiovisual</b> | <b>Video resolution</b>                       | <b>Video codec</b>                                                                    | <b>Video type</b>       |
+|---------------------------|-------------------------------|--------------------------|-----------------------------------------------|---------------------------------------------------------------------------------------|-------------------------|
+|                           | ITU-T J.246                   | Video                    | QCIF, CIF , VGA                               | H.264/AVC (MPEG-4 Part 10), VC-1, Windows Media 9, Real Video (RV 10), MPEG-4 Part 2. |                         |
+|                           | ITU-T J.341                   | Video                    | 1080i,1080p,1080i 50 Hz.1080p                 | H.264/AVC (MPEG-4 Part 10), MPEG-2                                                    |                         |
+| TCP                       | ITU-T P.1201 Amd. 2, App. III | Audiovisual              | HVGA, HD (1080i50, 1080p24, 1080i60, 1080p30) | H.264 (MPEG-4 Part 10)                                                                | Progressive downloading |
+
+# Appendix I
+
+(This appendix does not form an integral part of this Recommendation.)
+
+Table I.1 provides information on existing and planned QoE assessment standards to give the users of this Recommendation an overall view of existing and ongoing projects in ITU. In addition, Table I.2 and Table I.3 provide indication of existing and planned QoE assessment standards for video streaming applications.
+
+**Table I.1 – Existing Recommendations and current work in ITU on QoE assessment**
+
+| Application     | Media             | Conversational (CONV)/Non-conversational (NONCONV) | Subjective test methodology                                      | Objective test methodology                                                                |                    |          |               |
+|-----------------|-------------------|----------------------------------------------------|------------------------------------------------------------------|-------------------------------------------------------------------------------------------|--------------------|----------|---------------|
+|                 |                   |                                                    |                                                                  | Model                                                                                     | Transport protocol | FR/RR/NR | Primary usage |
+| Telephony       | Speech            | NONCONV                                            | [ITU-T P.800]<br>[ITU-T P.806]<br>[ITU-T P.830]<br>[ITU-T P.835] | [ITU-T P.862] + [ITU-T P.862.1] (NB)<br>[ITU-T P.862.2] (WB)<br>[ITU-T P.863] (NB/WB/SWB) |                    | FR       | LAB, MON      |
+|                 |                   |                                                    | [ITU-T P.1301]<br>[ITU-T P.1302]<br>[ITU-T P.1311]               | [ITU-T P.563] (NB)<br>[ITU-T P.564] (NB/WB)                                               |                    | NR       | MON           |
+|                 |                   | CONV                                               | [ITU-T P.800]<br>[ITU-T P.805]<br>[ITU-T P.1301]                 | [ITU-T G.107] (NB)<br>[ITU-T G.107.1] (WB)                                                |                    | NR       | PLN           |
+|                 |                   |                                                    | [ITU-T P.1312]                                                   | [ITU-T P.561] + [ITU-T P.562] (NB/WB)                                                     |                    | NR       | MON           |
+| Video telephony | Multimedia (Note) | NONCONV                                            | [ITU-T P.1302]                                                   |                                                                                           |                    |          |               |
+|                 |                   | CONV                                               | [ITU-T P.920]<br>[ITU-T P.1301]                                  | [ITU-T G.1070] (NB/WB)                                                                    |                    | NR       | PLN           |
+
+**Table I.1 – Existing Recommendations and current work in ITU on QoE assessment**
+
+| Application                      | Media      | Conversational (CONV)/Non-conversational (NONCONV) | Subjective test methodology                                                | Objective test methodology                                                                                                                                                                      |                    |          |               |
+|----------------------------------|------------|----------------------------------------------------|----------------------------------------------------------------------------|-------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------|--------------------|----------|---------------|
+|                                  |            |                                                    |                                                                            | Model                                                                                                                                                                                           | Transport protocol | FR/RR/NR | Primary usage |
+| Video streaming (Mobile TV/IPTV) | Video      | NONCONV                                            | [ITU-T P.910]<br>[ITU-T J.140]<br>[ITU-R BT.500-13]                        | [ITU-T J.144] standard definition (SD)<br>[ITU-T J.247] (QCIF, common intermediate format (CIF), video graphics array (VGA))<br>[ITU-T J.341] (HD)                                              | Transport agnostic | FR       | LAB, MON      |
+|                                  |            |                                                    |                                                                            | [ITU-T J.343.5] (HD, encrypted bitstream)<br>[ITU-T J.343.6] (HD, non-encrypted bitstream)                                                                                                      | UDP                | FR       | LAB, MON      |
+|                                  |            |                                                    |                                                                            | [ITU-T J.249] (SD)<br>[ITU-T J.246] (QCIF, CIF, VGA)<br>[ITU-T J.342] (HD)                                                                                                                      | Transport agnostic | RR       | MON           |
+|                                  |            |                                                    |                                                                            | [ITU-T J.343.3] (HD, encrypted bitstream)<br>[ITU-T J.343.4] (HD, non-encrypted bitstream)                                                                                                      | UDP                | RR       | MON           |
+|                                  |            |                                                    |                                                                            | [ITU-T J.343.1] (HD, encrypted bitstream)<br>[ITU-T J.343.2] (HD, non-encrypted bitstream)                                                                                                      | UDP                | NR       | MON           |
+|                                  | Audio      | NONCONV                                            | [ITU-T P.830]<br>[ITU-R BS.1116-3]<br>[ITU-R BS.1285]<br>[ITU-R BS.1534-3] | [ITU-R BS.1387-1]                                                                                                                                                                               |                    | FR/RR    | MON/PLN       |
+|                                  | Multimedia | NONCONV                                            | [ITU-T P.911]                                                              | [ITU-T P.1201.1] (QCIF, quarter video graphics array (QVGA), half video graphics array (HVGA))<br>[ITU-T P.1201.2] (SD, HD)<br>[ITU-T P.1202.1] (QCIF, QVGA, HVGA)<br>[ITU-T P.1202.2] (SD, HD) | UDP                | NR       | MON           |
+|                                  |            |                                                    |                                                                            | [ITU-T P.1201 Amd. 2], App. III (HD)<br>P.NATS phase1(SD,HD)<br>P.NATS phase2 (SD,HD,WQHD,UHD)                                                                                                  | TCP                | NR       | MON           |
+|                                  |            |                                                    |                                                                            | [ITU-T G.1071] (SD, HD)<br>G.OM_HEVC (SD,HD)                                                                                                                                                    | UDP                | NR       | PLN           |
+| Web browsing                     | Data       |                                                    | [ITU-T P.1501]                                                             | [ITU-T G.1030]                                                                                                                                                                                  | TCP                | NR       | PLN           |
+
+NOTE – For individual media (i.e., speech and video), the Recommendations used in telephony and video-streaming applications are applicable.
+
+**Table I.2 – Summary of existing and ongoing planning models for video streaming applications**
+
+| Transport protocol | Model                               | Video/Audiovisual | Video resolution                                                          | Video codec                              | Video type |
+|--------------------|-------------------------------------|-------------------|---------------------------------------------------------------------------|------------------------------------------|------------|
+| UDP                | ITU-T G.1071<br>(Lower resolution)  | Video/Audiovisual | HVGA, QVGA, QCIF                                                          | MPEG-4 Part 2, H.264<br>(MPEG-4 Part 10) |            |
+|                    | ITU-T G.1071<br>(Higher resolution) | Video/Audiovisual | SD (PAL/NTSC),<br>HD (720p50, 720p60, 1080i50, 1080p25, 1080i60, 1080p30) | H.264 (MPEG-4 Part 10)                   |            |
+|                    | G.O.M_HEVC                          | Video/Audiovisual | Phase1: HD                                                                | Phase1: ITU-T H.265<br>(MPEG-H Part 2)   | IPTV       |
+| TCP                |                                     |                   |                                                                           |                                          |            |
+
+**Table I.3 – Summary of existing and ongoing monitoring models for video streaming applications**
+
+| Transport protocol | Model                         | Video/Audiovisual | Video resolution                                                          | Video codec                                                                                                 | Video type                           |
+|--------------------|-------------------------------|-------------------|---------------------------------------------------------------------------|-------------------------------------------------------------------------------------------------------------|--------------------------------------|
+| UDP                | ITU-T P.1201.1                | Audiovisual       | HVGA, QVGA, QCIF                                                          | MPEG-4 Part 2, H.264<br>(MPEG-4 Part 10)                                                                    | IPTV                                 |
+|                    | ITU-T P.1201.2                | Audiovisual       | SD (PAL/NTSC),<br>HD (720p50, 720p60, 1080i50, 1080p25, 1080i60, 1080p30) | H.264 (MPEG-4 Part 10)                                                                                      | IPTV                                 |
+|                    | ITU-T J.343                   | Video             | HD, VGA/WVGA                                                              | H.264/AVC (MPEG-4 Part 10)                                                                                  | IPTV                                 |
+|                    | ITU-T J.247                   | Video             | QCIF, CIF, VGA                                                            | H.264/AVC (MPEG-4 part 10),<br>VC-1, Windows Media 9, Real Video (RV 10), MPEG-4 Part 2                     |                                      |
+|                    | ITU-T J.246                   | Video             | QCIF, CIF, VGA                                                            | H.264/AVC (MPEG-4 Part 10),<br>VC-1, Windows Media 9, Real Video (RV 10), MPEG-4 Part 2                     |                                      |
+|                    | ITU-T J.341                   | Video             | 1080i, 1080p, 1080i 50 Hz, 1080p                                          | H.264/AVC (MPEG-4 Part 10), MPEG-2                                                                          |                                      |
+| TCP                | ITU-T P.1201 Amd. 2, App. III | Audiovisual       | HVGA, HD (1080i50, 1080p24, 1080i60, 1080p30)                             | ITU-T H.264 (MPEG-4 Part 10)                                                                                | Progressive downloading              |
+|                    | ITU-T P.NATS phase1           | Audiovisual       | 240p, 360p, 480p, 720p, 1080p                                             | ITU-T H.264 (MPEG-4 Part 10) baseline, main, high, and high10 profiles                                      | Progressive and adaptive downloading |
+|                    | P.NATS phase2                 | Audiovisual       | 240p, 360p, 480p, 720p, 1080p, 2K, 4K                                     | ITU-T H.264 (MPEG-4 Part 10) baseline, main, high, and high10 profiles<br>H.265 baseline, main, high<br>VP9 | Progressive and adaptive downloading |
+
+# Appendix II
+
+(This appendix does not form an integral part of this Recommendation.)
+
+In this appendix, common metrics are defined for QoE assessment models for video streaming applications, which could be transmitted via TCP or UDP.
+
+## II.1 Metrics related to TCP based streaming video
+
+The following table defines common metrics for buffering in TCP based streaming video. These metrics could be used in OTT streaming video transmitted via HTTP/TCP, HLS/HTTP/TCP, DASH/HTTP/TCP.
+
+**Table II.1 – Definition for buffering in TCP based streaming video**
+
+| Metrics           | Definition                        |
+|-------------------|-----------------------------------|
+| Initial buffering | Reference to definition in G.102y |
+| Re-buffering      | Reference to definition in G.102y |
+| Stalling          | Reference to definition in G.102y |
+
+## II.2 Metrics related to UDP based streaming video
+
+The following table defines common metrics for buffering in UDP based streaming video. These metrics could be used in streaming video transmitted via UDP, such as IPTV.
+
+**Table II.2 – Definition for buffering in UDP based streaming video**
+
+| Metrics           | Definition                                  |
+|-------------------|---------------------------------------------|
+| Initial buffering | Reference to definition in [b-ITU-T G.1021] |
+| Re-buffering      | Reference to definition in [b-ITU-T G.1021] |
+
+# Bibliography
+
+- [b-ITU-T G.1021] Recommendation ITU-T G.1021 (2012), *Buffer models for development of client performance metrics*.
+- [b-ITU-T P.10 Amd.2] Recommendation ITU-T P.10/G.100 (2006) Amendment 2 (2008), *New definitions for inclusion in Recommendation ITU-T P.10/G.100*.
+- [b-DSL Forum] DSL Forum TR-126 (2006), *Triple-play Services Quality of Experience (QoE) Requirements*.
+- [b-Guilford, 1954] Guilford, J.P. (1954), *Psychometric Methods*, 2nd edition. New York, NY: McGraw Hill. 597 p.
+- [b-Jekosch, 2005] Jekosch, U. (2005), *Voice and speech quality perception – Assessment and evaluation*. Berlin: Springer. 208 p.
+- [b-Mattila, 2001] Mattila, V.V. (2001), *Descriptive analysis of speech quality in mobile communications*, Vol. 340. Doctoral dissertation thesis, Tampere University of Technology, Tampere. See also AES E-Library, [paper No. 5455](#) (viewed 2016-09-30)  
+<<http://www.aes.org/e-lib/browse.cfm?elib=9880>>
+- [b-Möller, 2000] Möller, S. (2000), *Assessment and prediction of speech quality in telecommunications*, Boston, MA: Kluwer. 244 p.
+- [b-Möller, 2002] Möller, S., Raake, A. (2002). Telephone speech quality prediction: Towards network planning and monitoring models. *Speech Communication Vol. 38*: pp. 47-75  
+<<http://www.sciencedirect.com/science/article/pii/S0167639301000437>>
+- [b-Möller, 2005] Möller, S. (2005), *Quality of telephone-based spoken dialogue systems*. New York, NY: Springer. 469 p.
+- [b-Quackenbush, 1988] Quackenbush, S.R., Barnwell III, T.P., Clements, M.A. (1988). *Objective measures of speech quality*. Englewood Cliffs, NJ: Prentice Hall. 377 p.
+- [b-Stevens, 1951] Stevens, S.S. (1951). *Stevens' handbook of experimental psychology*. New York, NY: Wiley. 1436 p.
+
+
+
+
+
+## **SERIES OF ITU-T RECOMMENDATIONS**
+
+|                 |                                                                                                                                                           |
+|-----------------|-----------------------------------------------------------------------------------------------------------------------------------------------------------|
+| Series A        | Organization of the work of ITU-T                                                                                                                         |
+| Series D        | General tariff principles                                                                                                                                 |
+| Series E        | Overall network operation, telephone service, service operation and human factors                                                                         |
+| Series F        | Non-telephone telecommunication services                                                                                                                  |
+| <b>Series G</b> | <b>Transmission systems and media, digital systems and networks</b>                                                                                       |
+| Series H        | Audiovisual and multimedia systems                                                                                                                        |
+| Series I        | Integrated services digital network                                                                                                                       |
+| Series J        | Cable networks and transmission of television, sound programme and other multimedia signals                                                               |
+| Series K        | Protection against interference                                                                                                                           |
+| Series L        | Environment and ICTs, climate change, e-waste, energy efficiency; construction, installation and protection of cables and other elements of outside plant |
+| Series M        | Telecommunication management, including TMN and network maintenance                                                                                       |
+| Series N        | Maintenance: international sound programme and television transmission circuits                                                                           |
+| Series O        | Specifications of measuring equipment                                                                                                                     |
+| Series P        | Terminals and subjective and objective assessment methods                                                                                                 |
+| Series Q        | Switching and signalling                                                                                                                                  |
+| Series R        | Telegraph transmission                                                                                                                                    |
+| Series S        | Telegraph services terminal equipment                                                                                                                     |
+| Series T        | Terminals for telematic services                                                                                                                          |
+| Series U        | Telegraph switching                                                                                                                                       |
+| Series V        | Data communication over the telephone network                                                                                                             |
+| Series X        | Data networks, open system communications and security                                                                                                    |
+| Series Y        | Global information infrastructure, Internet protocol aspects and next-generation networks, Internet of Things and smart cities                            |
+| Series Z        | Languages and general software aspects for telecommunication systems                                                                                      |

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+
+
+**ITU-T**
+
+TELECOMMUNICATION  
+STANDARDIZATION SECTOR  
+OF ITU
+
+**G.1021**
+
+(07/2012)
+
+SERIES G: TRANSMISSION SYSTEMS AND MEDIA,  
+DIGITAL SYSTEMS AND NETWORKS
+
+Multimedia Quality of Service and performance – Generic  
+and user-related aspects
+
+# --- **Buffer models for development of client performance metrics**
+
+Recommendation ITU-T G.1021
+
+## ITU-T G-SERIES RECOMMENDATIONS **TRANSMISSION SYSTEMS AND MEDIA, DIGITAL SYSTEMS AND NETWORKS**
+
+|                                                                                                                                              |                      |
+|----------------------------------------------------------------------------------------------------------------------------------------------|----------------------|
+| INTERNATIONAL TELEPHONE CONNECTIONS AND CIRCUITS                                                                                             | G.100–G.199          |
+| GENERAL CHARACTERISTICS COMMON TO ALL ANALOGUE CARRIER-TRANSMISSION SYSTEMS                                                                  | G.200–G.299          |
+| INDIVIDUAL CHARACTERISTICS OF INTERNATIONAL CARRIER TELEPHONE SYSTEMS ON METALLIC LINES                                                      | G.300–G.399          |
+| GENERAL CHARACTERISTICS OF INTERNATIONAL CARRIER TELEPHONE SYSTEMS ON RADIO-RELAY OR SATELLITE LINKS AND INTERCONNECTION WITH METALLIC LINES | G.400–G.449          |
+| COORDINATION OF RADIOTELEPHONY AND LINE TELEPHONY                                                                                            | G.450–G.499          |
+| TRANSMISSION MEDIA AND OPTICAL SYSTEMS CHARACTERISTICS                                                                                       | G.600–G.699          |
+| DIGITAL TERMINAL EQUIPMENTS                                                                                                                  | G.700–G.799          |
+| DIGITAL NETWORKS                                                                                                                             | G.800–G.899          |
+| DIGITAL SECTIONS AND DIGITAL LINE SYSTEM                                                                                                     | G.900–G.999          |
+| <b>MULTIMEDIA QUALITY OF SERVICE AND PERFORMANCE – GENERIC AND USER-RELATED ASPECTS</b>                                                      | <b>G.1000–G.1999</b> |
+| TRANSMISSION MEDIA CHARACTERISTICS                                                                                                           | G.6000–G.6999        |
+| DATA OVER TRANSPORT – GENERIC ASPECTS                                                                                                        | G.7000–G.7999        |
+| PACKET OVER TRANSPORT ASPECTS                                                                                                                | G.8000–G.8999        |
+| ACCESS NETWORKS                                                                                                                              | G.9000–G.9999        |
+
+*For further details, please refer to the list of ITU-T Recommendations.*
+
+## Recommendation ITU-T G.1021
+
+# Buffer models for development of client performance metrics
+
+## Summary
+
+Recommendation ITU-T G.1021 defines buffer models that describe the behaviour of the client buffers used at the receiver side of audio/video applications.
+
+Buffer models receive as input IP packets associated with meta information on their content and provide as output client states associated with relevant timing information.
+
+The buffer models are intended to be used for the development of client performance metrics in order to complement the IP-layer metrics defined in Recommendation ITU-T Y.1540.
+
+## History
+
+| Edition | Recommendation | Approval   | Study Group |
+|---------|----------------|------------|-------------|
+| 1.0     | ITU-T G.1021   | 2012-07-14 | 12          |
+
+## FOREWORD
+
+The International Telecommunication Union (ITU) is the United Nations specialized agency in the field of telecommunications, information and communication technologies (ICTs). The ITU Telecommunication Standardization Sector (ITU-T) is a permanent organ of ITU. ITU-T is responsible for studying technical, operating and tariff questions and issuing Recommendations on them with a view to standardizing telecommunications on a worldwide basis.
+
+The World Telecommunication Standardization Assembly (WTSA), which meets every four years, establishes the topics for study by the ITU-T study groups which, in turn, produce Recommendations on these topics.
+
+The approval of ITU-T Recommendations is covered by the procedure laid down in WTSA Resolution 1.
+
+In some areas of information technology which fall within ITU-T's purview, the necessary standards are prepared on a collaborative basis with ISO and IEC.
+
+## NOTE
+
+In this Recommendation, the expression "Administration" is used for conciseness to indicate both a telecommunication administration and a recognized operating agency.
+
+Compliance with this Recommendation is voluntary. However, the Recommendation may contain certain mandatory provisions (to ensure, e.g., interoperability or applicability) and compliance with the Recommendation is achieved when all of these mandatory provisions are met. The words "shall" or some other obligatory language such as "must" and the negative equivalents are used to express requirements. The use of such words does not suggest that compliance with the Recommendation is required of any party.
+
+## INTELLECTUAL PROPERTY RIGHTS
+
+ITU draws attention to the possibility that the practice or implementation of this Recommendation may involve the use of a claimed Intellectual Property Right. ITU takes no position concerning the evidence, validity or applicability of claimed Intellectual Property Rights, whether asserted by ITU members or others outside of the Recommendation development process.
+
+As of the date of approval of this Recommendation, ITU had not received notice of intellectual property, protected by patents, which may be required to implement this Recommendation. However, implementers are cautioned that this may not represent the latest information and are therefore strongly urged to consult the TSB patent database at <http://www.itu.int/ITU-T/ipr/>.
+
+© ITU 2013
+
+All rights reserved. No part of this publication may be reproduced, by any means whatsoever, without the prior written permission of ITU.
+
+## Table of Contents
+
+|                                                                                              | Page |
+|----------------------------------------------------------------------------------------------|------|
+| 1 Scope .....                                                                                | 1    |
+| 2 References.....                                                                            | 2    |
+| 3 Definitions .....                                                                          | 2    |
+| 3.1 Terms defined elsewhere .....                                                            | 2    |
+| 4 Abbreviations and acronyms .....                                                           | 2    |
+| 5 Conventions .....                                                                          | 3    |
+| 6 De-jitter buffers .....                                                                    | 3    |
+| 6.1 Network jitter and de-jitter buffers.....                                                | 3    |
+| 6.2 De-jitter buffers for audio applications .....                                           | 5    |
+| 6.3 De-jitter buffers for "low bitrate mode" and "high bitrate mode" video applications..... | 5    |
+| 6.4 De-jitter buffers for over-the-top video applications.....                               | 6    |
+| 6.5 De-jitter buffer performance outcomes .....                                              | 7    |
+| 7 De-jitter buffer models .....                                                              | 7    |
+| 7.1 Maximum buffer size .....                                                                | 9    |
+| 7.2 Service strategy .....                                                                   | 9    |
+| 7.3 Initial buffering strategy .....                                                         | 10   |
+| 7.4 Re-buffering strategy .....                                                              | 11   |
+| Annex A – Pseudocode of the state-machine description of the de-jitter buffer model .....    | 13   |
+| A.1 Algorithm overview.....                                                                  | 13   |
+| A.2 Configuration parameters .....                                                           | 13   |
+| A.3 De-jitter buffer variables .....                                                         | 13   |
+| A.4 De-jitter buffer input.....                                                              | 14   |
+| A.5 De-jitter buffer high level function pseudocode.....                                     | 14   |
+| Bibliography.....                                                                            | 17   |
+
+
+
+# Buffer models for development of client performance metrics
+
+# 1 Scope
+
+Audio and video applications currently used in packet networks implement client buffers (e.g., de-jitter buffers and play-out buffers) at the receiver side. The behaviour of these buffers is essential to the end-to-end performance of these applications. This Recommendation defines models for these client buffers for audio/video applications currently used in actual networks, in order to estimate the client state based on the dynamic network behaviour.
+
+In this Recommendation, different buffer models are derived depending on the different applications taken into account. However, these buffer models may be based on common intrinsic characteristics: i.e., buffer size, service rate, initial delay and re-buffering delay. Example buffer performance metrics include peak/average/instaneous buffer occupation.
+
+In this Recommendation, buffer models may accept as input, either live traffic arriving on measurement points, packet capture (pcap) files, or post-processed files. Buffer models are expected to provide client states (e.g., codec play-out begin, buffer overflow, buffer dry out, play-out resume) associated with relevant timing information. As part of future work, these buffer models could be used to derive new performance parameters and associated statistics in order to complement the IP-layer metrics defined in Recommendation Y.1540.
+
+It is out of scope for this Recommendation to define specific de-jitter buffers for use in actual client media stream delivery.
+
+In this Recommendation the following applications, rate and buffer characteristics, and associated protocol stacks are considered:
+
+- a) Audio applications:
+  - i) Ideally constant fill, constant play-out rate and relatively low bandwidth.
+- b) Video "low bitrate mode" quarter common intermediate format-quarter video graphics array (QCIF-QVGA) resolutions, mostly for mobile TV and streaming with the sub application areas:
+  - i) Linear mobile TV over real-time transport protocol (RTP) (including mobile TV over a 3G mobile network with multimedia broadcast/multicast services (MBMS) and with unicast transport over RTP/UDP/IP).
+  - ii) Multimedia streaming (including 3GPP packet-switched streaming service (PSS) with transport over RTP/UDP/IP).
+  - iii) [IETF RFC 3984] variable bit rate, up to 2 Mbps, with separate buffers for audio and video, and separate play-out rates for each buffer.
+- c) Video "high bitrate mode" standard-definition (SD) and high-definition (HD) television, mostly for Internet Protocol television (IPTV) with the following sub-application areas:
+  - i) Linear broadcast TV (including transmission over MPEG2-TS/RTP/UDP/IP, MPEG2-TS/UDP/IP and RTP/UDP/IP transport).
+  - ii) Video on-demand (including transmission over MPEG2-TS/RTP/UDP/IP, MPEG2-TS/UDP/IP and RTP/UDP/IP transport).
+  - iii) [IETF RFC 2250], constant bit rate (CBR) or variable bit rate (VBR) up to 50 Mbps, with separate buffers for audio and video, and separate play-out rates for each buffer.
+
+- d) Over-the-top video applications:
+  - i) Non-adaptive streaming (MPEG4\_fileformat/HTTP/TCP, HTTP progressive download, HTTP partial download, chunks):
+    - Progressive download: TCP transport, TCP flow control (65Mbps super-peak rates, peak rate depends on network conditions and path), classic single buffer, leaky bucket at constant frame rate.
+  - ii) Adaptive streaming dynamic adaptive streaming over HTTP (DASH/TCP):
+    - Chunk Delivery: (e.g., DASH) bursts of packets, like sequential FTP file transfers.
+
+# 2 References
+
+The following ITU-T Recommendations and other references contain provisions which, through reference in this text, constitute provisions of this Recommendation. At the time of publication, the editions indicated were valid. All Recommendations and other references are subject to revision; users of this Recommendation are therefore encouraged to investigate the possibility of applying the most recent edition of the Recommendations and other references listed below. A list of the currently valid ITU-T Recommendations is regularly published. The reference to a document within this Recommendation does not give it, as a stand-alone document, the status of a Recommendation.
+
+- [ITU-T G.1020] Recommendation ITU-T G.1020 (2006), *Performance parameter definitions for quality of speech and other voiceband applications utilizing IP networks*.
+- [ITU-T H.222.0] Recommendation ITU-T H.222.0 (2006) | ISO/IEC 13818-1:2007, *Information technology – Generic coding of moving pictures and associated audio information: Systems*.
+- [ITU-T Y.1540] Recommendation ITU-T Y.1540 (2010), *Internet protocol data communication service – IP packet transfer and availability performance parameters*.
+- [IETF RFC 2250] IETF RFC 2250 (1998), *RTP Payload Format for MPEG1/MPEG2 Video*.
+- [IETF RFC 3984] IETF RFC 3984 (2005), *RTP Payload Format for H.264 Video*.
+
+# 3 Definitions
+
+## 3.1 Terms defined elsewhere
+
+This Recommendation uses the following terms defined elsewhere:
+
+**3.1.1 decoding time-stamp (DTS)** [ITU-T H.222.0]: A field that may be present in a PES packet header that indicates the time that an access unit is decoded in the system target decoder.
+
+# 4 Abbreviations and acronyms
+
+This Recommendation uses the following abbreviations and acronyms:
+
+|      |                                       |
+|------|---------------------------------------|
+| ARQ  | Automatic Retransmission on Request   |
+| CBR  | Constant Bit Rate                     |
+| DASH | Dynamic Adaptive Streaming over HTTP  |
+| DSL  | Digital Subscriber Line               |
+| DSP  | Digital Signal Processor              |
+| DTS  | Decoding Time Stamp                   |
+| EWMA | Exponentially Weighted Moving Average |
+
+|         |                                         |
+|---------|-----------------------------------------|
+| HTTP    | Hypertext Transfer Protocol             |
+| IP      | Internet Protocol                       |
+| IPTV    | Internet Protocol Television            |
+| FEC     | Forward Error Correction                |
+| FTP     | File Transfer Protocol                  |
+| MBMS    | Multimedia Broadcast/Multicast Services |
+| MPEG    | Moving Picture Experts Group            |
+| MPEG-TS | MPEG Transport Stream                   |
+| NTP     | Network Time Protocol                   |
+| OTT     | Over The Top                            |
+| PCAP    | Packet Capture                          |
+| PCR     | Program Clock Reference                 |
+| PDV     | Packet Delay Variation                  |
+| PES     | Packetized Elementary Stream            |
+| PLC     | Packet Loss Concealment                 |
+| PLL     | Phase-Locked Loop                       |
+| PSS     | Packet-switched Streaming Service       |
+| QCIF    | Quarter Common Intermediate Format      |
+| QVGA    | Quarter Video Graphics Array            |
+| RTP     | Real-time Transport Protocol            |
+| STB     | Set-Top Box                             |
+| VBR     | Variable Bit Rate                       |
+
+# 5 Conventions
+
+The buffer model behavior is described by the state-machine pseudocode provided in Annex A of this Recommendation.
+
+Pseudocode is an informal, high-level description of the operating principle of an algorithm. It uses the structural conventions of a programming language, but is intended for human reading rather than machine reading. Pseudocode typically omits details that are not essential for human understanding of the algorithm, such as variable declarations, system-specific code and some subroutines.
+
+Note that no standard for pseudocode syntax exists. In this Recommendation, pseudo-code similar to the C programming language has been used.
+
+# 6 De-jitter buffers
+
+## 6.1 Network jitter and de-jitter buffers
+
+In the context of computer networks, the term "jitter" is often used as a measure of the variability over time of packet latency across a network. A network with constant latency has no variation and therefore no jitter. However, the word is imprecise and, in the context of packet-switched networks, the term "packet delay variation" is often preferred over "jitter" (see [ITU-T Y.1540]).
+
+In order to smooth out the natural packet delay variation or jitter effects of asynchronous networks and to synchronize play-out between sender and receiver, most techniques introduce an additional delay, implemented by having the packets arriving on the reception side equipment being temporarily stored in a buffer. This buffer is designed to counteract the jitter that is introduced by the network, until the moment the audio signal or the image is delivered to the decoding scheme.
+
+This buffer is called the "de-jitter buffer." In order to ensure a continuous play-out of streaming audio or video, the de-jitter buffer parameters have to be tuned, such that, at the moment an audio or a picture is to be played-out, the entire content of the audio or picture resides in the buffer.
+
+There is a trade-off between the end-to-end delay caused by the de-jitter buffer and the packet loss. Packets that arrive before timeout are played out in the same order as they were created at the sender side. Those packets that arrive after timeout are discarded. An ideal algorithm should neither delay the play-out time excessively, nor discard too many packets.
+
+As shown in Figure 6-1, the end-to-end delay experienced by an IP packet (carrying speech or voice band applications) can be decomposed in several parts: packetization delay, network transfer delay, de-jitter buffer delay (accommodating delay variation from source and network) and play-out buffer delay.
+
+![Figure 6-1: Delay of packet networks and network elements. This Gantt-style diagram illustrates the timeline of delays for an IP packet. The horizontal axis represents time, and the vertical axis represents the packet's progress. The diagram is divided into four main sections by vertical dashed lines: 'Source delays', 'Network transfer delay', 'De-jitter buffer', and 'Other destination delays'. 'Source delays' include 'Packetization' (a rectangle) and 'Minimum additional source delay' (an arrow). 'Network transfer delay' includes 'Minimum transfer delay' (a rectangle) and '99.9%-tile transfer delay' (a dashed rectangle). 'De-jitter buffer' is a rectangle labeled 'Accommodates delay variation from network and source terminal'. 'Other destination delays' include 'Minimum/ playout buffer' and 'DSP and other queuing' (both rectangles). The diagram is labeled 'G.1021(12)_F6-1' at the bottom right.](daa4a6fa7e2ba1954258f86b4928eb32_img.jpg)
+
+Figure 6-1: Delay of packet networks and network elements. This Gantt-style diagram illustrates the timeline of delays for an IP packet. The horizontal axis represents time, and the vertical axis represents the packet's progress. The diagram is divided into four main sections by vertical dashed lines: 'Source delays', 'Network transfer delay', 'De-jitter buffer', and 'Other destination delays'. 'Source delays' include 'Packetization' (a rectangle) and 'Minimum additional source delay' (an arrow). 'Network transfer delay' includes 'Minimum transfer delay' (a rectangle) and '99.9%-tile transfer delay' (a dashed rectangle). 'De-jitter buffer' is a rectangle labeled 'Accommodates delay variation from network and source terminal'. 'Other destination delays' include 'Minimum/ playout buffer' and 'DSP and other queuing' (both rectangles). The diagram is labeled 'G.1021(12)\_F6-1' at the bottom right.
+
+NOTE – Figure 6-1 is an exact replica of Figure 5 of [ITU-T G.1020].
+
+**Figure 6-1 – Delay of packet networks and network elements**
+
+This type of decomposition can be generalized to most audio/video applications.
+
+### 6.1.1 Fixed size de-jitter buffers
+
+The size of the play-out buffer may be kept fixed; this scheme is referred to as fixed size play-out buffering, or static scheduling. The end-to-end delay is kept constant for all packets in a session. If packets exceed a given transmission time they are discarded.
+
+Although the fixed method, which uses a fixed buffer size, is easy to implement, it can result in unsatisfactory audio or video quality because there is no optimal delay when network conditions vary with time. The fluctuating end-to-end delays experienced in IP networks may cause delay to increase to a level that may be annoying to users (when the buffer is too large), or may cause packet losses due to their late arrival times (when the buffer is too small).
+
+### 6.1.2 Adaptive size de-jitter buffers
+
+The size of the play-out buffer may also be adaptively adjusted during the transmission; this scheme is referred to as adaptive size play-out buffering, or dynamic scheduling.
+
+Adaptive techniques adapt the delay to changing network conditions, thus preventing the use of long delays during low-congestion conditions and vice versa. In this case, the network behavior is monitored and network delays are continuously estimated and the play-out delay is dynamically adjusted.
+
+## 6.2 De-jitter buffers for audio applications
+
+Figure 6-2 indicates the positions of measurement points and system components for speech and voiceband applications utilizing IP networks. As shown, the de-jitter buffer is placed after processing the IP and RTP/UDP layers.
+
+![Figure 6-2: Destination terminal components for audio applications. The diagram shows two parallel processing paths. The top path represents the audio signal flow: De-jitter buffer -> De-packetize -> Decoder and PLC -> D/A converter -> Spkr. Reference points are marked: 'Ear reference point' at the speaker, 'Receive electrical reference point' at the D/A converter output, and 'Terminal output reference point' at the physical interface input. The bottom path represents the network packet flow: Physical interface -> Lower layers -> IP -> RTP/UDP. Arrows indicate the flow of Waveform (solid green), Continuous digital (dashed blue), and Asynchronous packets (solid orange).](5a4e62bead259c258d069fd3663ea670_img.jpg)
+
+The diagram illustrates the signal flow and reference points for audio applications in a destination terminal. It consists of two main horizontal paths. The upper path shows the audio signal processing: starting from the 'Terminal output reference point' at the 'Physical interface', the signal (represented by an orange arrow for asynchronous packets) flows through 'Lower layers', 'IP', and 'RTP/UDP' blocks. It then enters the 'De-jitter buffer', followed by 'De-packetize', 'Decoder and PLC', and 'D/A converter' blocks. The signal (now a green waveform arrow) exits the 'D/A converter' at the 'Receive electrical reference point' and is played by the 'Spkr' (speaker). The 'Ear reference point' is indicated at the speaker. The lower path shows the network packet flow: starting from the 'Physical interface', the signal flows through 'Lower layers', 'IP', and 'RTP/UDP' blocks. A legend at the bottom indicates: 'Waveform' (solid green arrow), 'Continuous digital' (dashed blue arrow), and 'Asynchronous packets' (solid orange arrow). The diagram is labeled 'G.1021(12)\_F6-2'.
+
+Figure 6-2: Destination terminal components for audio applications. The diagram shows two parallel processing paths. The top path represents the audio signal flow: De-jitter buffer -> De-packetize -> Decoder and PLC -> D/A converter -> Spkr. Reference points are marked: 'Ear reference point' at the speaker, 'Receive electrical reference point' at the D/A converter output, and 'Terminal output reference point' at the physical interface input. The bottom path represents the network packet flow: Physical interface -> Lower layers -> IP -> RTP/UDP. Arrows indicate the flow of Waveform (solid green), Continuous digital (dashed blue), and Asynchronous packets (solid orange).
+
+NOTE – Figure 6-2 is an exact replica of Figure 3 of [ITU-T G.1020].
+
+**Figure 6-2 – Destination terminal components for audio applications**
+
+For audio applications, the main objective of de-jitter buffers is to keep the packet loss rate under a given threshold (e.g., 5 per cent) and to keep the end-to-end delay as small as possible.
+
+Audio applications usually take advantage of the signal structure alternating talkspurts and silent periods. With adaptive methods, play-out adjustment is performed during the silent periods between talkspurts. The adjustment is done on the first packet of the talkspurt; all packets in the same talkspurt are scheduled to play out at fixed intervals following the play-out of the first packet. This mechanism uses the same play-out delay throughout a given talkspurt, but permits different play-out delays for different talkspurts. The variation of the play-out delay may introduce artificially elongated or reduced silence periods, but such modification of silence periods is considered acceptable in the perceived speech if that variation is reasonably limited.
+
+If the jitter is so large that it causes packets to be received out of the range of this buffer, the out-of-range packets are discarded and dropouts are heard in the audio. Usually packet loss concealment (PLC) features are employed together with the decoder in order to mask the possible effects of packet losses. One PLC method is artificial regeneration of the packet received prior to the lost packet, followed by insertion of the duplicated packet into the gap. In another method, the digital value of the dropped packet is estimated by interpolation and an artificially generated packet is inserted on that basis. However, when jitter exceeds what the digital signal processor (DSP) can handle to make up for the missing packets, the quality of service (QoS) of audio is degraded.
+
+## 6.3 De-jitter buffers for "low bitrate mode" and "high bitrate mode" video applications
+
+The decomposition of system components can also be developed for Video "low bitrate mode" and Video "high bitrate mode" applications that are also based on UDP, as shown in Figure 6-3.
+
+![Figure 6-3: Destination terminal components for 'low bitrate mode' and 'high bitrate mode' video applications. The diagram shows the flow of data from a terminal input reference point through various processing stages to a speaker and video set. The flow starts at the 'Terminal input reference point', passes through 'Physical interface', 'Lower layers', 'IP', and 'RTP/UDP'. A 'Buffer input reference point' is located between 'IP' and 'RTP/UDP'. The data then enters a 'De-jitter buffer shared with FEC/ARQ', followed by 'De-packetize'. A 'Buffer output reference point' is located between 'De-packetize' and the decoders. The data is then split into 'Decoder and PLC audio' and 'Decoder and PLC video', which are followed by 'D/A converter' and finally 'Speaker' and 'Video set' respectively. A legend at the bottom indicates: Waveform (green solid line), Continuous digital (blue dashed line), and Asynchronous packets (orange solid line). The diagram is labeled G.1021(12)_F6-3.](042733dc5e8e7f5f30b60adba3266cde_img.jpg)
+
+```
+graph LR; TIRP[Terminal input reference point] --> PI[Physical interface]; PI --> LL[Lower layers]; LL --> IP[IP]; IP --> RU[RTP/UDP]; RU --> DB[De-jitter buffer shared with FEC/ARQ]; DB --> DP[De-packetize]; DP --> DPA[Decoder and PLC audio]; DP --> DPV[Decoder and PLC video]; DPA --> DAC[D/A converter]; DAC --> S[Speaker]; DPV --> DVAC[D/A converter]; DVAC --> VS[Video set]; BIRP[Buffer input reference point] -.-> IP; BORP[Buffer output reference point] -.-> DPA; BORP -.-> DPV;
+```
+
+Figure 6-3: Destination terminal components for 'low bitrate mode' and 'high bitrate mode' video applications. The diagram shows the flow of data from a terminal input reference point through various processing stages to a speaker and video set. The flow starts at the 'Terminal input reference point', passes through 'Physical interface', 'Lower layers', 'IP', and 'RTP/UDP'. A 'Buffer input reference point' is located between 'IP' and 'RTP/UDP'. The data then enters a 'De-jitter buffer shared with FEC/ARQ', followed by 'De-packetize'. A 'Buffer output reference point' is located between 'De-packetize' and the decoders. The data is then split into 'Decoder and PLC audio' and 'Decoder and PLC video', which are followed by 'D/A converter' and finally 'Speaker' and 'Video set' respectively. A legend at the bottom indicates: Waveform (green solid line), Continuous digital (blue dashed line), and Asynchronous packets (orange solid line). The diagram is labeled G.1021(12)\_F6-3.
+
+**Figure 6-3 – Destination terminal components for "low bitrate mode" and "high bitrate mode" video applications**
+
+In the case of video applications, techniques for combating transmission errors have been developed along two directions.
+
+In the first direction, traditional error control and recovery schemes for data communications have been extended for video transmission. These application-layer loss correction techniques aim at lossless recovery. Examples of such schemes include forward error correction (FEC) and automatic retransmission on request (ARQ). Usually FEC or ARQ mechanisms are performed together with de-jitter recovery in the same functional block.
+
+In the second direction, signal-reconstruction and error-concealment techniques have been proposed that strive to obtain a close approximation of the original signal, or attempt to make the output signal at the decoder the least objectionable to human eyes. However, unlike data transmissions where lossless delivery is absolutely required, human eyes can tolerate a certain degree of distortion in image and video signals. For instance, video encoders such as ITU-T H.264 may insert resynchronization markers in the compression loop as well as a structural bitstream arrangement to allow the decoder to continue the decoding process despite erasures encountered in the compressed image, which essentially prevents frame skipping.
+
+If the FEC source block structure is chosen with proper alignment, for example with the random access points of the video stream, then the FEC latency can be absorbed into the video decoding buffer latency. This would mean that the protection period is in general not additive to the end-to-end latency, but generally lower.
+
+## 6.4 De-jitter buffers for over-the-top video applications
+
+Video delivered over the Internet without involvement of the service provider's own video service (but on the public Internet) is defined as over-the-top (OTT).
+
+The delivery mode may be based on non-adaptive streaming (e.g., HTTP progressive download), where transport is accomplished using TCP flow control (the peak rate depending on network conditions), and where play-out is performed at constant frame rate.
+
+It may also be based on adaptive streaming (DASH/TCP), which consists of delivering chunks (i.e., bursts of packets) like sequential file transfers. Well-known implementations include: Microsoft smooth streaming, HTTP live streaming by Apple, and Adobe dynamic streaming.
+
+The decomposition in system components for OTT video applications may be adapted as shown in Figure 6-4.
+
+![Figure 6-4: Destination terminal components for OTT video applications. The diagram shows the flow of data from applications through various processing stages to output devices. The flow starts with Apps (http, rtmp, ...) sending data to a Container/demux. The Container/demux sends data to two parallel paths: one for audio (De-jitter buffer for audio -> Decoder and PLC audio -> D/A converter -> Speaker) and one for video (De-jitter buffer for video -> Decoder and PLC video -> D/A converter -> Video set). A Buffer output reference point is indicated at the output of the decoders. The data also flows through a Physical interface, Lower layers, IP, and TCP. A Terminal input reference point is indicated at the Physical interface, and a Buffer input reference point is indicated at the IP layer. A legend at the bottom indicates that green arrows represent Waveform, dashed blue arrows represent Continuous digital, and orange arrows represent Asynchronous packets.](eefe19c5e14dc4d6c316b7f7fbb7d7d7_img.jpg)
+
+```
+
+graph LR
+    Apps["Apps (http, rtmp, ...)"] -- "Asynchronous packets" --> Container["Container/demux"]
+    Container -- "Continuous digital" --> DeJitterAudio["De-jitter buffer for audio"]
+    Container -- "Continuous digital" --> DeJitterVideo["De-jitter buffer for video"]
+    DeJitterAudio -- "Continuous digital" --> DecoderAudio["Decoder and PLC audio"]
+    DeJitterVideo -- "Continuous digital" --> DecoderVideo["Decoder and PLC video"]
+    DecoderAudio -- "Waveform" --> DACAudio["D/A converter"]
+    DecoderVideo -- "Waveform" --> DACVideo["D/A converter"]
+    DACAudio -- "Waveform" --> Speaker["Speaker"]
+    DACVideo -- "Waveform" --> VideoSet["Video set"]
+    Container -- "Asynchronous packets" --> PhysIntf["Physical interface"]
+    PhysIntf -- "Asynchronous packets" --> LowerLayers["Lower layers"]
+    LowerLayers -- "Asynchronous packets" --> IP["IP"]
+    IP -- "Asynchronous packets" --> TCP["TCP"]
+    TCP -- "Asynchronous packets" --> Apps
+    BufferOutput["Buffer output reference points"] -.-> DecoderAudio
+    BufferOutput -.-> DecoderVideo
+    TerminalInput["Terminal input reference point"] -.-> PhysIntf
+    BufferInput["Buffer input reference point"] -.-> IP
+  
+```
+
+G.1021(12)\_F6-4
+
+Legend:   
+ → Waveform (green solid line)  
+ - - - Continuous digital (blue dashed line)  
+ → Asynchronous packets (orange solid line)
+
+Figure 6-4: Destination terminal components for OTT video applications. The diagram shows the flow of data from applications through various processing stages to output devices. The flow starts with Apps (http, rtmp, ...) sending data to a Container/demux. The Container/demux sends data to two parallel paths: one for audio (De-jitter buffer for audio -> Decoder and PLC audio -> D/A converter -> Speaker) and one for video (De-jitter buffer for video -> Decoder and PLC video -> D/A converter -> Video set). A Buffer output reference point is indicated at the output of the decoders. The data also flows through a Physical interface, Lower layers, IP, and TCP. A Terminal input reference point is indicated at the Physical interface, and a Buffer input reference point is indicated at the IP layer. A legend at the bottom indicates that green arrows represent Waveform, dashed blue arrows represent Continuous digital, and orange arrows represent Asynchronous packets.
+
+**Figure 6-4 – Destination terminal components for OTT video applications**
+
+## 6.5 De-jitter buffer performance outcomes
+
+In the case of UDP-based applications, the performance of a de-jitter buffer may be characterized by two types of outcomes corresponding to the following undesirable situations:
+
+- The de-jitter buffer overflows, which corresponds to a new packet arriving in a full de-jitter buffer which is unable to store it; this new packet is lost,
+- The de-jitter buffer dries out, which corresponds to an empty de-jitter buffer upon play-out request by the decoder.
+
+In the case of OTT applications, the performance of a de-jitter buffer is different because the TCP protocol adapts the stream bitrate according to network conditions. Therefore, the performance will be characterized by the following outcomes: the buffer is playing, the buffer is stopped, or the buffer is buffering or re-buffering.
+
+When the buffer is buffering, play-out devices usually display a black screen with an icon showing that a buffering or re-buffering process is taking place. For re-buffering the last displayed frame is typically displayed.
+
+# 7 De-jitter buffer models
+
+Even if all manufacturers use de-jitter buffers in the reception side equipment, many different technical implementations may exist, depending on manufacturers, on applications (video or audio) and on the protocols used (RTP/UDP, HTTP, etc.).
+
+A general description of the de-jitter buffer model is shown in Figure 7-1.
+
+![Diagram of the De-jitter buffer model. A central box labeled 'De-jitter buffer model' receives three inputs: 'Configuration parameters' from the top, 'IP packets + meta info' from the left, and a feedback signal from the bottom. The feedback signal is labeled 'G.1021(12)_F7-1'. The model produces an output labeled '[Time, buffer state]' to the right.](4801720824e4b5e2361a5564f91cfb70_img.jpg)
+
+```
+
+graph TD
+    CP[Configuration parameters] --> DJBM[De-jitter buffer model]
+    IP[IP packets + meta info] --> DJBM
+    DJBM --> OS["[Time, buffer state]"]
+    OS -- G.1021(12)_F7-1 --> DJBM
+  
+```
+
+Diagram of the De-jitter buffer model. A central box labeled 'De-jitter buffer model' receives three inputs: 'Configuration parameters' from the top, 'IP packets + meta info' from the left, and a feedback signal from the bottom. The feedback signal is labeled 'G.1021(12)\_F7-1'. The model produces an output labeled '[Time, buffer state]' to the right.
+
+**Figure 7-1 – De-jitter buffer model**
+
+The de-jitter buffer has four main configuration parameters: maximum buffer size, service strategy, initial buffering size and re-buffering size.
+
+The **maximum buffer size** is the amount of memory dedicated to the de-jitter buffer.
+
+The **service strategy** describes how packets or frames are played out by the application.
+
+The **initial buffering size**, also called build-up time, is the amount of audio or video data of a stream which are stored in the buffer before playing out the first packet of this stream in order to minimize the risk of buffer dry-out.
+
+The **re-buffering size** is the amount of audio or video data of a stream which are stored in the buffer before resuming play-out of the first packet of a flow, after the buffer has dried out, in order to minimize the risk that the buffer will dry out again immediately. Note that a similar configuration description is also used by the Broadband Forum Technical Report [b-BBF-TR-135], which defines in the case of video applications, set-top box (STB) characteristics such as:
+
+- MaxDe-jitteringBufferSize: describes the maximum de-jittering buffer size, in bytes, supported by the STB.
+- BufferSize: de-jittering buffer size, measured in milliseconds, on the receiver side of the customer premises equipment (CPE). The STB is responsible for converting this into the actual buffer size in bytes.
+- BufferInitialLevel: describes the number of milliseconds worth of data that should be in the receiver side de-jittering buffer before play-out can start. Typically, this would be half of the buffer size.
+
+The input to the de-jitter buffer model is composed of IP packets relative to the stream of interest, their arrival times, and meta information on their content relative to this stream as follows:
+
+- offset of the different audio/video components contained in the packet;
+- information relative to the frames of the audio component (frame counter, frame size, DTS, is the frame complete or not);
+- information relative to the frames of the video component (frame counter, frame size, DTS, is the frame complete or not).
+
+The output of the de-jitter buffer model is the state of the de-jitter buffer (i.e., playing, stopped, buffering, or re-buffering), together with the time of entry in each state.
+
+This behaviour may be represented by the state machine described in Figure 7-2.
+
+![State-machine description of the de-jitter buffer model. The diagram shows five states: Initial buffering, Missing, Playing, Rebuffering, and Stopped. Transitions are as follows: Initial buffering leads to Playing and Stopped. Playing leads to Missing, Rebuffering, and Stopped. Missing leads to Rebuffering. Rebuffering leads to Playing and Stopped. Stopped has a self-loop and leads to Initial buffering. The diagram is labeled G.1021(12)_F7-2.](33ed1f9b27c7c21c797aa928b0f06851_img.jpg)
+
+```
+
+stateDiagram-v2
+    [*] --> Initial buffering
+    Initial buffering --> Playing
+    Initial buffering --> Stopped
+    Playing --> Missing
+    Playing --> Rebuffering
+    Playing --> Stopped
+    Missing --> Rebuffering
+    Rebuffering --> Playing
+    Rebuffering --> Stopped
+    Stopped --> Stopped
+    Stopped --> Initial buffering
+  
+```
+
+State-machine description of the de-jitter buffer model. The diagram shows five states: Initial buffering, Missing, Playing, Rebuffering, and Stopped. Transitions are as follows: Initial buffering leads to Playing and Stopped. Playing leads to Missing, Rebuffering, and Stopped. Missing leads to Rebuffering. Rebuffering leads to Playing and Stopped. Stopped has a self-loop and leads to Initial buffering. The diagram is labeled G.1021(12)\_F7-2.
+
+**Figure 7-2 – State-machine description of the de-jitter buffer model**
+
+In the model, the state of the de-jitter buffer is estimated at time epochs corresponding to each packet arrival and to each frame extraction to the decoder. A pseudo-code description of this model is given in Annex A of this Recommendation.
+
+## 7.1 Maximum buffer size
+
+The maximum buffer size depends on the type of application.
+
+Note that the maximum buffer size is usually expressed in milliseconds (ms). However, it can easily be expressed in bytes when the average bit rate of the application is known.
+
+For audio applications (e.g., VoIP) generating IP packets every 20 ms, the buffer generally may contain only a small number of packets (up to two packets), thus corresponding to 40 ms.
+
+For "high bitrate mode" video applications (based on MPEG2) generating IP packets every 4 ms on average, the buffer generally may contain a larger number of packets; a maximum buffer size of 80 ms is commonly used, corresponding to 20 packets in this case.
+
+For OTT video applications (e.g., progressive download, DASH), the buffer generally may contain a very large number of packets. Classical implementations are currently using maximum buffer sizes of several tens of seconds.
+
+## 7.2 Service strategy
+
+The service strategy describes the way packets are played out by the decoder. Packets or frames are usually played out using a constant service rate, at least on relatively long periods of time. However, the service strategy may be more complex.
+
+### 7.2.1 UDP-based applications
+
+The service rate generally depends on the service clock used by the application server sending the real-time application packets (e.g., IPTV server). In an ideal case, the service rate matches the service clock (i.e., the decoder's clock matches the encoder's system clock).
+
+Two buffer handling methods, push and pull, are commonly used. In the push method, the buffer controls the flow of data and periodically transfers appropriate data to the decoder. In the pull method, the decoder requests data from the buffer and the buffer transfers the requested data to the decoder.
+
+Several options may be used for the service strategy:
+
+- *The service rate is independent of the service clock used by the application server*; it can be based, for example, on a local free running oscillator (e.g., VoIP applications). In this case, depending on the offset between the clocks used at both sides, and on the duration of the
+
+communication, some effects can be noticed on the de-jitter buffer. It should be noted that, even when the service rate follows the service clock used by the application server, the use of a free running clock is generally required at the beginning of a clock recovery process during the stabilization period.
+
+- *The service rate follows the service clock used by the application server, which is synchronized by a synchronization protocol* (e.g., network time protocol (NTP) or IEEE 1588). Such a mechanism may be accurate enough to ensure that no significant drift between service rate and service clock is observed during the service lifetime.
+- *The service rate follows the service clock used by the application server, which is recovered at the de-jitter buffer side.* This clock recovery can be obtained either from the timestamps contained in the application packets, or from the knowledge of the nominal rate of the application (in case of constant bit rate flows). In general, the clock recovery is not perfect, as it is impacted by the packet delay variation of the network (e.g., low frequency components of the network packet delay variation (PDV) may not be filtered by the clock recovery mechanism).
+
+Adaptive de-jitter strategies, such as adaptation of the play-out rate when the buffer is almost empty in an effort to avoid de-jitter buffer starvation, are often included as well. The adaptation logic is based on the jitter estimates computed from the arrival characteristics of the media packets. Adaptive de-jittering involves introducing discontinuities in the media play-out, which may be offensive to the listener or viewer.
+
+UDP-based flows usually carry their own clock structure. For example, RTP flows and MPEG video flows contain service timestamps. In RTP flows, IP packets contain a 4-byte timestamp that is used to enable the receiver to play back the received samples at appropriate intervals. In addition, this feature generalizes the use of this protocol to variable bitrate flows.
+
+In MPEG2 video flows, MPEG2 frame information is encapsulated within MPEG2 transport stream (MPEG2-TS) packets, which are in turn transported in IP packets, according to the MPEG/RTP/UDP/IPv4 protocol stack. MPEG2-TS flows are organized in streams related to the different components of the audio-visual application (video, audio, etc.). Each stream generates 188-byte packet elementary stream (PES) packets. These PES packets are then assembled in groups of seven, resulting in 1316-byte IP packets ( $7 \times 188 = 1316$ ). PES packets may contain a program clock reference (PCR) enabling the decoder to present synchronized content, such as audio tracks matching the associated video. Usually the PCR is imbedded in the video PES packets.
+
+The bit rates outputted from MPEG-2 encoders may be constant bit rate (CBR) or variable bit rate (VBR). CBR MPEG-2 streams operate at a constant bandwidth data throughput regardless of the complexity of the video content while VBR MPEG-2 streams are encoded using different bit rates. The CBR technique is the predominant mode of operation for carrying video services over digital subscriber line (DSL) based networks.
+
+If a service clock is available in IP packets, the clock reference timestamps, which are samples of the system time clock, are applicable both to a decoder and to an encoder. Therefore, they can be utilized to implement clock reconstruction control loops in decoders with sufficient accuracy. Annex D in [ITU-T H.222.0] gives some suggestions for implementing them.
+
+Another method of slaving the decoder's clock to the received data stream is via a phase-locked loop (PLL). Classical PLL implementations are based on exponentially weighted moving average (EWMA) methods, where the weight is chosen in order to smooth the PLL reaction.
+
+### 7.2.2 OTT applications
+
+OTT applications typically use constant play-out rates for the audio and the video components. Typical play-out rates for the audio component are 44100 samples/s grouped in multiple samples per frame.
+
+Typical play-out rates for the video component are either 24 frames/s or 30 frames/s, depending on the standard used for TV sets.
+
+## **7.3 Initial buffering strategy**
+
+### **7.3.1 UDP-based applications**
+
+The initial buffering size added to the very first packet of a given flow must be chosen carefully since it impacts the performance of the application.
+
+When small buffers are used (e.g., conversational audio services), the initial buffering size is constrained by the maximum buffer size itself and three main buffering strategies can be used.
+
+- The first strategy assumes that the de-jitter buffer is full when media assessment begins. The monitored transmissions are allowed to consume the complete buffer resource before any buffer dry-out event is registered. In this strategy, the probability of a packet experiencing a buffer overflow becomes very high.
+- The second strategy assumes that the de-jitter buffer is empty when media assessment begins, the opposite of the first approach. The monitored transmissions are allowed to completely fill-out the buffer before any buffer overflow event is registered. In this strategy, the probability of a packet experiencing a buffer dry-out increases. This strategy is often used in VoIP applications since the overall delay is the key parameter.
+- The third strategy assumes that the de-jitter buffer is half-full when media assessment begins. The monitored transmissions are allowed to consume half the buffer resource before any buffer dry-out event is registered, or to completely fill-out the buffer before any buffer overflow event is registered. This strategy minimizes the probability of a packet experiencing a buffer overflow or a buffer dry-out, and it is commonly used in video applications.
+
+When performing repeated/continuous stream monitoring operations, it should be permissible to pass the buffer occupation information from one assessment interval to the next. This would permit longer-term impairment (re-buffering) occurrence to be assessed while still satisfying the need for short-term results. More importantly, re-buffering will likely have a drastically negative effect on user experience and therefore, long-term assessment is seen as valuable.
+
+### **7.3.2 OTT applications**
+
+OTT applications commonly use large buffers.
+
+The choice of the initial buffering size is generally a compromise between the time a user has to wait before the video is displayed on the screen, and the probability of re-buffering events.
+
+Current implementations use initial buffering sizes ranging from 2 seconds to 10 seconds.
+
+## **7.4 Re-buffering strategy**
+
+### **7.4.1 UDP-based applications**
+
+Before defining a re-buffering strategy, it is necessary to detect whether re-buffering is needed. Several criteria may be used for deciding when re-buffering is needed.
+
+- When a sufficiently long sequence of packets arrives too late for their play-out time, these packets will all be designated as discarded. In practice, any combination of consecutive lost and discarded packets in sufficient number should cause re-buffering.
+- The variable number of consecutive lost or discarded packets can be matched to the specific receivers in use. A default value can be set, following discussion.
+
+- Alternatively, a threshold could be set on average buffer occupation (delay) over a sliding window of packets, and trigger re-buffering when the average falls below a variable threshold (again, with a default value).
+
+When a de-jitter buffer dries out during media assessment, different strategies are possible, depending on the origin of the event.
+
+- If the event occurs as the de-jitter buffer is slowly drying out, this likely corresponds to a slow drift between the sender clock and the receiver clock. This drift should normally be taken into account by the de-jitter buffer service rate. However, this mechanism may fail to prevent the de-jitter buffer to dry out. Also, the bitrate available in the network may be less than the average encoder rate; this produces the same effect. In this situation, there is no way for the de-jitter buffer to refill naturally. Therefore, the solution should be to consider that the first packet observed after de-jitter buffer dry-out is the first packet of a new period, that the buffer has to be considered in an initial state, and that the media assessment is interrupted (i.e., a video freeze or an audio silence).
+- If the de-jitter buffer is depleted rapidly, this likely corresponds to a severe degradation of network conditions, producing a combination of consecutive lost packets (packets lost by the network) and discarded packets (packets arriving too late for their play-out time) in sufficient numbers. In this situation, the de-jitter buffer may refill naturally when the period of consecutive lost and discarded packets ends. But the natural refill of the buffer may take some time, during which a dry-out event is more likely to happen. Therefore, in this situation also, the solution could be to consider that the first packet observed after de-jitter buffer dry-out is the first packet of a new period, that the buffer is in an initial state, and that the media assessment is interrupted (corresponding to an unavailable period).
+
+Therefore, in all cases the re-buffering delay chosen may be equal to the initial buffering delay.
+
+### 7.4.2 OTT applications
+
+In the case of OTT applications, no information is lost at the application layer since TCP compensates for poor network conditions by managing packet retransmissions in order to guarantee loss-free behavior to the application. The side effect of the possible packet retransmissions is that the de-jitter buffer may dry out and generate a re-buffering event.
+
+The re-buffering size may be different than the initial buffering size. Furthermore, if several re-buffering outcomes occur, the sizes of these different re-buffering outcomes may not be identical. Most play-out devices adapt to varying network conditions. If the performance of the network degrades further, it is likely that the player will increase the re-buffering size in order to avoid additional buffer dry-out and re-buffering outcomes.
+
+# Annex A
+
+## Pseudocode of the state-machine description of the de-jitter buffer model
+
+(This annex forms an integral part of this Recommendation.)
+
+Annex A provides the pseudocode associated to the state-machine describing the behavior of the de-jitter buffer.
+
+## A.1 Algorithm overview
+
+The de-jitter buffer receives input packets that have been processed by RTP, HTTP, MPEG-TS, Flash Video (FLV), MP4 and other application layer protocols which identify the locations of media frames in the packets. This information is conveyed to the de-jitter buffer with each packet as a media frame information list. The de-jitter buffer AddPacket is called each time a new packet arrives, and RemoveMediaFrame is called each time a media frame is scheduled to be decoded. In the absence of a model that can call the de-jitter buffer to remove media frames, it can be simply be modeled by a play-out process with the use of a timer, which calls the RemoveMediaFrame function at a configurable, fixed time interval directly associated with the frame rate used for playback in the application area.
+
+The output of the model includes a list of events recorded by the calls to the record event function.
+
+The StopNotification function can be called externally to stop the operation of de-jitter buffer.
+
+## A.2 Configuration parameters
+
+*Maximum buffer duration:* maximum duration of media frames in the de-jitter buffer
+
+*Initial buffering duration:* minimum duration of media frames in the de-jitter buffer needed before the model enters play state at the start of the session
+
+*Re-buffering duration:* minimum duration of media frames in the de-jitter buffer needed before the model enters play state after having entered the re-buffering state
+
+*Drop buffer duration:* minimum duration of media frames in the de-jitter buffer needed before the model enters play state after having entered the missing state
+
+*Missing packet wait duration:* maximum duration that this buffer model is allowed to spend in the missing state
+
+*Blocking mode:* boolean needed for applications such as OTT where the buffer model is not allowed to drop frames. When set to 1 and the buffer fills up, the buffer model will not drop frames and will return a failure condition, indicating to the rest of the system that the packet was not added and needs to be reinserted. When set to 0 and the buffer fills up, the buffer model will drop frames and will return a success condition, indicating to the rest of the system that the packet was processed.
+
+## A.3 De-jitter buffer variables
+
+*Current state:* the state of the de-jitter buffer one of: (stopped, playing, initial buffering, re-buffering, missing), initialized to initial buffering
+
+*Next DTS:* the expected DTS of the next media frame to be played, initialized to the DTS of the first valid media frame received
+
+*Time buffered:* the cumulative duration of all the frames stored in the de-jitter buffer, initialized to 0
+
+*Dropped packet count:* the number of packets that have been dropped, initialized to 0
+
+*Discarded packets*: a list of packets (packet objects) that have been discarded because the buffer was full, initially this list is empty
+
+*Buffered packets*: a list of packets (packet objects) that have been received but not modeled as played, initially this list is empty
+
+## A.4 De-jitter buffer input
+
+*Packet*: Describes the input of the de-jitter buffer model. This can be called a packet object and it contains the IP packet including a media frame information list. The media frame information list is described as follows:
+
+- If an IP packet does not contain media frame data, the media frame information list is empty. Example media frame information list: {}
+- If an IP packet contains part of one media frame, the media frame information list contains one entry composed of (IP packet arrival time, byte offset to the start of media frame data in the IP packet, the size in bytes of this media frame data, the type of media: audio or video, boolean indicating this is a part of a media frame, DTS in milliseconds of the frame). Example media frame information list: {12:01:32.001, 34, 101, video, 0 for partial, 00:00:00.000}
+- If an IP packet contains one media frame, the media frame information list contains one entry composed of (IP packet arrival time, byte offset to the start of media frame data in the IP packet, the size in bytes of this media frame data, the type of media: audio or video, boolean indicating this is a complete a media frame, DTS in milliseconds of the media frame). Example media frame information: {12:01:32.001, 34, 101, video, 1 for complete, 00:00:00.000}
+- If an IP packet contains multiple media frames then the media frame information list contains one entry for each complete or incomplete media frame, in the order that they appear in the packet. Example media frame information list: { {12:01:32.001, 34, 101, video, 0, 00:00:00.000}, {12:01:32.001, 135, 50, video, 1, 00:00:00.041}, {12:01:32.001, 185,100, video, 0, 00:00:00.083} }
+
+## A.5 De-jitter buffer high level function pseudocode
+
+```
+Boolean AddPacket(Packet)
+    IF (current state == Stopped) THEN
+        record event()
+        return success
+    END IF
+
+    IF (Packet DTS < Next DTS) THEN
+        dropped packet count ++
+        record event()
+        return success
+    END IF
+
+    IF (current state == Playing) THEN
+        IF (Time Buffered > Maximum Buffer Duration) THEN
+            IF (Blocking Mode) THEN
+                return failure
+            ELSE
+                dropped packet count ++
+                insert (in DTS order) Packet to Discarded Packets
+                record event()
+                return success
+            END IF
+        END IF
+    END IF
+END IF
+```
+
+```
+
+    insert (in DTS order) Packet to Buffered Packets
+    FOR (each Media Frame completed by, or contained in Packet)
+        Time Buffered += Media Frame Duration
+    END FOR
+
+    IF (current state == Initial Buffering) THEN
+        IF (Time Buffered > Initial Buffering Duration) THEN
+            current state = Playing
+        END IF
+    ELSE IF (current state == Re Buffering) THEN
+        head DTS = Get earliest DTS of a complete Media frame in Buffered
+Packets
+        IF (Time Buffered > Re Buffering Duration) THEN
+            IF (head DTS <= next DTS) THEN
+                current state = Playing
+            ELSE
+                missing state start time = current time
+                current state = Missing
+            END IF
+        END IF
+    ELSE IF (current state == Missing) THEN
+        head DTS = Get earliest DTS of a complete Media frame in Buffered
+Packets
+        IF (head DTS <= next DTS) THEN
+            current state = Playing
+        END IF
+        IF ((Time Buffered > Drop Buffer Duration) or
+            ((current time - missing state start time) > missing packet wait
+duration)) THEN
+            next DTS = head DTS
+            current state = Playing
+        END IF
+    END IF
+
+    record event()
+    return success
+
+StopNotification()
+    current state = Stopped
+
+RemoveMediaFrame()
+    IF (current state == Stopped) THEN
+        record event()
+        return
+    END IF
+
+    IF (current state == Missing) THEN
+        IF ((current time - missing state start time) > missing packet wait
+duration) THEN
+            head DTS = Get earliest DTS of a complete media frame in Buffered
+Packets
+
+            next DTS = head DTS
+            current state = Playing
+        END IF
+    END IF
+
+    IF (current state == Playing) THEN
+        buffered head DTS = Get earliest DTS of a complete media frame in
+Buffered Packets
+        dropped head DTS = Get earliest DTS of media frame in Dropped Packets
+        IF (buffered head DTS <= next DTS) THEN
+            removed Frame = remove next (in DTS order) complete Media frame
+from Buffered Packets
+
+```
+
+```
+
+        remove any partial frames with earlier DTS than next DTS from
+Buffered Packets
+        next DTS = removed Frame DTS + removed Frame Duration
+        Time Buffered -= removed Frame Duration
+        ELSE IF (dropped head DTS <= next DTS) THEN
+            removed Frame = remove next (in DTS order) Media frame from Dropped
+Packets
+            next DTS = removed Frame DTS + removed Frame Duration
+        ELSE
+            current state = Rebuffering
+        END IF
+    END IF
+    record event()
+
+record event()
+    generate a message indicating the current state including:
+```
+
+- Current time
+- Current state
+- Next DTS
+- Time Buffered
+- Dropped packet count
+- Number of packets in Buffered Packets
+- Number of packets in Dropped Packets
+
+# Bibliography
+
+- [b-BBF-TR-135] Broadband Forum Technical Report (2007), *Data Model for a TR-069 Enabled STB*.
+
+
+
+
+
+## SERIES OF ITU-T RECOMMENDATIONS
+
+|                 |                                                                                             |
+|-----------------|---------------------------------------------------------------------------------------------|
+| Series A        | Organization of the work of ITU-T                                                           |
+| Series D        | General tariff principles                                                                   |
+| Series E        | Overall network operation, telephone service, service operation and human factors           |
+| Series F        | Non-telephone telecommunication services                                                    |
+| <b>Series G</b> | <b>Transmission systems and media, digital systems and networks</b>                         |
+| Series H        | Audiovisual and multimedia systems                                                          |
+| Series I        | Integrated services digital network                                                         |
+| Series J        | Cable networks and transmission of television, sound programme and other multimedia signals |
+| Series K        | Protection against interference                                                             |
+| Series L        | Construction, installation and protection of cables and other elements of outside plant     |
+| Series M        | Telecommunication management, including TMN and network maintenance                         |
+| Series N        | Maintenance: international sound programme and television transmission circuits             |
+| Series O        | Specifications of measuring equipment                                                       |
+| Series P        | Terminals and subjective and objective assessment methods                                   |
+| Series Q        | Switching and signalling                                                                    |
+| Series R        | Telegraph transmission                                                                      |
+| Series S        | Telegraph services terminal equipment                                                       |
+| Series T        | Terminals for telematic services                                                            |
+| Series U        | Telegraph switching                                                                         |
+| Series V        | Data communication over the telephone network                                               |
+| Series X        | Data networks, open system communications and security                                      |
+| Series Y        | Global information infrastructure, Internet protocol aspects and next-generation networks   |
+| Series Z        | Languages and general software aspects for telecommunication systems                        |

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+
+
+International Telecommunication Union
+
+**ITU-T**
+
+TELECOMMUNICATION  
+STANDARDIZATION SECTOR  
+OF ITU
+
+**G.1028.2**
+
+(06/2019)
+
+SERIES G: TRANSMISSION SYSTEMS AND MEDIA,  
+DIGITAL SYSTEMS AND NETWORKS
+
+Multimedia Quality of Service and performance – Generic  
+and user-related aspects
+
+---
+
+**Assessment of the LTE circuit switched fall  
+back – Impact on voice quality of service**
+
+Recommendation ITU-T G.1028.2
+
+ITU-T
+
+![ITU logo: International Telecommunication Union](1d7527f4316cfe2d342b08d1653d1592_img.jpg)
+
+The logo of the International Telecommunication Union (ITU) features a stylized globe with a red lightning bolt striking across it. The letters 'ITU' are prominently displayed in blue, with the full name 'International Telecommunication Union' written in smaller blue text to the right.
+
+ITU logo: International Telecommunication Union
+
+# ITU-T G-SERIES RECOMMENDATIONS **TRANSMISSION SYSTEMS AND MEDIA, DIGITAL SYSTEMS AND NETWORKS**
+
+|                                                                                                                                              |                      |
+|----------------------------------------------------------------------------------------------------------------------------------------------|----------------------|
+| INTERNATIONAL TELEPHONE CONNECTIONS AND CIRCUITS                                                                                             | G.100–G.199          |
+| GENERAL CHARACTERISTICS COMMON TO ALL ANALOGUE CARRIER-TRANSMISSION SYSTEMS                                                                  | G.200–G.299          |
+| INDIVIDUAL CHARACTERISTICS OF INTERNATIONAL CARRIER TELEPHONE SYSTEMS ON METALLIC LINES                                                      | G.300–G.399          |
+| GENERAL CHARACTERISTICS OF INTERNATIONAL CARRIER TELEPHONE SYSTEMS ON RADIO-RELAY OR SATELLITE LINKS AND INTERCONNECTION WITH METALLIC LINES | G.400–G.449          |
+| COORDINATION OF RADIOTELEPHONY AND LINE TELEPHONY                                                                                            | G.450–G.499          |
+| TRANSMISSION MEDIA AND OPTICAL SYSTEMS CHARACTERISTICS                                                                                       | G.600–G.699          |
+| DIGITAL TERMINAL EQUIPMENTS                                                                                                                  | G.700–G.799          |
+| DIGITAL NETWORKS                                                                                                                             | G.800–G.899          |
+| DIGITAL SECTIONS AND DIGITAL LINE SYSTEM                                                                                                     | G.900–G.999          |
+| <b>MULTIMEDIA QUALITY OF SERVICE AND PERFORMANCE – GENERIC AND USER-RELATED ASPECTS</b>                                                      | <b>G.1000–G.1999</b> |
+| TRANSMISSION MEDIA CHARACTERISTICS                                                                                                           | G.6000–G.6999        |
+| DATA OVER TRANSPORT – GENERIC ASPECTS                                                                                                        | G.7000–G.7999        |
+| PACKET OVER TRANSPORT ASPECTS                                                                                                                | G.8000–G.8999        |
+| ACCESS NETWORKS                                                                                                                              | G.9000–G.9999        |
+
+*For further details, please refer to the list of ITU-T Recommendations.*
+
+# Recommendation ITU-T G.1028.2
+
+# Assessment of the LTE circuit switched fall back – Impact on voice quality of service
+
+## Summary
+
+While the long term evolution (LTE) circuit switched fall back (CSFB) can be considered as a separate procedure from voice service delivery over LTE, it has an impact on the quality of service (QoS) for the voice services and is therefore worthy of attention and consideration. Indeed, operators have to make sure that clients purchasing LTE-capable devices must keep the level of QoS they are accustomed to when compared with their former 2G and 3G devices, even when VoLTE is unavailable (for any reason, such as, network or device). Therefore, specific and complementary key performance indicators (KPIs) are proposed in this Recommendation, together with procedures for their assessment.
+
+It should be noted that this Recommendation focuses only on the specific impact of CSFB. The information it contains must be considered as a complement to provisions already defined in Recommendation ITU-T G.1028.
+
+## History
+
+| Edition | Recommendation | Approval   | Study Group | Unique ID*                                                                |
+|---------|----------------|------------|-------------|---------------------------------------------------------------------------|
+| 1.0     | ITU-T G.1028.2 | 2019-06-29 | 12          | <a href="http://handle.itu.int/11.1002/1000/13928">11.1002/1000/13928</a> |
+
+## Keywords
+
+Circuit switched fall back, CSFB, LTE, quality of service, QoS, telephony, voice, VoLTE, 4G.
+
+---
+
+\* To access the Recommendation, type the URL <http://handle.itu.int/> in the address field of your web browser, followed by the Recommendation's unique ID. For example, <http://handle.itu.int/11.1002/1000/11830-en>.
+
+## FOREWORD
+
+The International Telecommunication Union (ITU) is the United Nations specialized agency in the field of telecommunications, information and communication technologies (ICTs). The ITU Telecommunication Standardization Sector (ITU-T) is a permanent organ of ITU. ITU-T is responsible for studying technical, operating and tariff questions and issuing Recommendations on them with a view to standardizing telecommunications on a worldwide basis.
+
+The World Telecommunication Standardization Assembly (WTSA), which meets every four years, establishes the topics for study by the ITU-T study groups which, in turn, produce Recommendations on these topics.
+
+The approval of ITU-T Recommendations is covered by the procedure laid down in WTSA Resolution 1.
+
+In some areas of information technology which fall within ITU-T's purview, the necessary standards are prepared on a collaborative basis with ISO and IEC.
+
+## NOTE
+
+In this Recommendation, the expression "Administration" is used for conciseness to indicate both a telecommunication administration and a recognized operating agency.
+
+Compliance with this Recommendation is voluntary. However, the Recommendation may contain certain mandatory provisions (to ensure, e.g., interoperability or applicability) and compliance with the Recommendation is achieved when all of these mandatory provisions are met. The words "shall" or some other obligatory language such as "must" and the negative equivalents are used to express requirements. The use of such words does not suggest that compliance with the Recommendation is required of any party.
+
+## INTELLECTUAL PROPERTY RIGHTS
+
+ITU draws attention to the possibility that the practice or implementation of this Recommendation may involve the use of a claimed Intellectual Property Right. ITU takes no position concerning the evidence, validity or applicability of claimed Intellectual Property Rights, whether asserted by ITU members or others outside of the Recommendation development process.
+
+As of the date of approval of this Recommendation, ITU had not received notice of intellectual property, protected by patents, which may be required to implement this Recommendation. However, implementers are cautioned that this may not represent the latest information and are therefore strongly urged to consult the TSB patent database at <http://www.itu.int/ITU-T/ipr/>.
+
+© ITU 2019
+
+All rights reserved. No part of this publication may be reproduced, by any means whatsoever, without the prior written permission of ITU.
+
+## Table of Contents
+
+|                                                        | Page |
+|--------------------------------------------------------|------|
+| 1 Scope.....                                           | 1    |
+| 2 References.....                                      | 1    |
+| 3 Definitions .....                                    | 1    |
+| 4 Abbreviations and acronyms .....                     | 1    |
+| 5 Conventions .....                                    | 2    |
+| 6 Overview of CSFB .....                               | 2    |
+| 7 KPIs for circuit switched fallback .....             | 5    |
+| 7.1 CSFB call setup failure ratio .....                | 6    |
+| 7.2 CSFB call set-up time.....                         | 6    |
+| 7.3 CSFB return to 4G/LTE time .....                   | 6    |
+| 7.4 CSFB return to 4G/LTE failure ratio.....           | 7    |
+| 7.5 Quality targets .....                              | 7    |
+| 8 Assessment methodology for CSFB .....                | 7    |
+| 8.1 Passive monitoring .....                           | 8    |
+| 8.2 Active monitoring.....                             | 8    |
+| 8.3 Drive testing .....                                | 8    |
+| 8.4 Trace files/other network related information..... | 8    |
+| 9 Impact on QoS .....                                  | 8    |
+| Bibliography.....                                      | 10   |
+
+# **Introduction**
+
+In early deployments of the fourth generation of mobile networks (LTE), it can happen that the delivery of multimedia services, and in particular voice over LTE (referred to in this Recommendation as VoLTE), is not possible by using a complete LTE infrastructure (with full IMS). This situation can remain applicable in the short and even medium term.
+
+In the meantime pending the deployment of VoLTE, operators must use the legacy networks (using 2G and 3G technologies) when there is a need for voice and SMS delivery to LTE users.
+
+The alternative to this issue is the use of CSFB, an interim solution allowing the usage of legacy networks, until full support for VoLTE is available.
+
+## Recommendation ITU-T G.1028.2
+
+## Assessment of the LTE circuit switched fall back – Impact on voice quality of service
+
+# 1 Scope
+
+This Recommendation provides guidance for assessment and measurement of the impact of LTE circuit switched fall back on QoS for voice services and provides means to express it in numerical terms, complementing provisions already defined in [ITU-T G.1028]. This Recommendation applies to voice service only.
+
+# 2 References
+
+The following ITU-T Recommendations and other references contain provisions which, through reference in this text, constitute provisions of this Recommendation. At the time of publication, the editions indicated were valid. All Recommendations and other references are subject to revision; users of this Recommendation are therefore encouraged to investigate the possibility of applying the most recent edition of the Recommendations and other references listed below. A list of the currently valid ITU-T Recommendations is regularly published. The reference to a document within this Recommendation does not give it, as a stand-alone document, the status of a Recommendation.
+
+[ITU-T G.1028] Recommendation ITU-T G.1028 (2019), *End-to-end quality of service for voice over 4G mobile networks*.
+
+[ETSI TR 103 219] ETSI TR 103 219 (2015), *Speech and multimedia Transmission Quality (STQ); Quality of Service aspects of voice communication in an LTE environment*.
+
+[ETSI TS 123 272] ETSI TS 123 272 (2018), *Digital cellular telecommunications system (Phase 2+) (GSM); Universal Mobile Telecommunications System (UMTS); LTE; Circuit Switched (CS) fallback in Evolved Packet System (EPS); Stage 2*.
+
+# 3 Definitions
+
+None.
+
+# 4 Abbreviations and acronyms
+
+This Recommendation uses the following abbreviations and acronyms:
+
+|         |                                                                              |
+|---------|------------------------------------------------------------------------------|
+| BSS     | Base Station System                                                          |
+| CS      | Circuit Switched                                                             |
+| CSFB    | Circuit Switched Fall Back                                                   |
+| E-UTRAN | Evolved Universal Mobile Telecommunication System Terrestrial Access Network |
+| GERAN   | GSM EDGE Radio Access Network                                                |
+| GPRS    | General Packet Radio Service                                                 |
+| GSM     | Global System for Mobile                                                     |
+| HO      | Hand Over                                                                    |
+| IMEI    | International Mobile Equipment Identity                                      |
+| IMS     | IP Multimedia Subsystem                                                      |
+
+|        |                                                           |
+|--------|-----------------------------------------------------------|
+| IMSI   | International Mobile Subscriber Identity                  |
+| KPI    | Key Performance Indicator                                 |
+| LCS    | Location Services                                         |
+| LTE    | Long Term Evolution of mobile telecommunications networks |
+| MME    | Mobility Management Entity                                |
+| MOS-LQ | Mean Opinion Score, Listening Quality                     |
+| MSC    | Mobile Switching Centre                                   |
+| NB     | Narrow Band                                               |
+| PGW    | Packet Data Network Gateway                               |
+| QoS    | Quality of Service                                        |
+| RAT    | Radio Access Technology                                   |
+| RF     | Radio Frequency                                           |
+| RRC    | Radio Resource Control                                    |
+| RNS    | Radio Network System                                      |
+| SGSN   | Serving GPRS Support Node                                 |
+| SGW    | Serving Gateway                                           |
+| SIB    | System Information Block                                  |
+| SMS    | Short Message Service                                     |
+| SWB    | Super Wideband                                            |
+| UDI    | Unique Device Identification                              |
+| UE     | User Equipment                                            |
+| UMTS   | Universal Mobile Telecommunications Service               |
+| USSD   | Unstructured Supplementary Service Data                   |
+| UTRAN  | UMTS Terrestrial Radio Access Network                     |
+| VoLTE  | Voice over LTE                                            |
+| WB     | Wide Band                                                 |
+
+# 5 Conventions
+
+None.
+
+# 6 Overview of CSFB
+
+CSFB is a set of procedures, defined by technological standards [ETSI TS 123 272], that allow UEs connected to LTE cellular networks without access to VoLTE services, to have access to circuit switched services (such as voice, CS UDI video, LCS, USSD) through a temporary (i.e., for the duration of the usage of the concerned service) fallback from E-UTRAN (4G) to UTRAN (3G) or GERAN (2G) circuit switched (CS) domains, assuming that an overlapping coverage of 4G and 3G/2G is available. This fallback means that the voice call is not active in LTE, but in 3G or 2G.
+
+Naturally, CSFB is only possible when users are using dual mode devices, i.e., able to operate both in LTE and UMTS or GSM networks.
+
+To support CSFB, specific interfaces are necessary to connect the equipment in charge of call processing in these technologies. In particular, Figure 1 shows SGs (connecting the mobility management entity (MME) to the mobile switching centre (MSC)) and S3 (between MME and serving GPRS support node (SGSN)) interfaces, i.e., the ones that allow a practical implementation of the CSFB procedure. These interfaces correspond to interface C3 in [ITU-T G.1028] (see clause 10.1.3).
+
+![Figure 1: Network interfaces required for handling of CSFB procedures. The diagram shows a User Equipment (UE) connected to three radio access networks: GERAN, UTRAN, and E-UTRAN. GERAN connects to an SGSN via Gb and A interfaces. UTRAN connects to an SGSN via Iu-ps and to an MSC via Iu-cs. E-UTRAN connects to an MME via S1-MME. The MME connects to the MSC via SGs and to the SGSN via S3. The SGSN connects to the MSC via Gs. The UE also has direct interfaces Um (to GERAN), Uu (to UTRAN), and LTE-Uu (to E-UTRAN).](acfc53eca625d62b38aa2563efa95c3e_img.jpg)
+
+```
+
+graph LR
+    UE[UE] -- Um --> GERAN[GERAN]
+    UE -- Uu --> UTRAN[UTRAN]
+    UE -- LTE-Uu --> E-UTRAN[E-UTRAN]
+    GERAN -- Gb --> SGSN[SGSN]
+    GERAN -- A --> SGSN
+    UTRAN -- Iu-ps --> SGSN
+    UTRAN -- Iu-cs --> MSC[MSC]
+    E-UTRAN -- S1-MME --> MME[MME]
+    MME -- SGs --> MSC
+    MME -- S3 --> SGSN
+    SGSN -- Gs --> MSC
+  
+```
+
+G.1028.2(19)\_F01
+
+Figure 1: Network interfaces required for handling of CSFB procedures. The diagram shows a User Equipment (UE) connected to three radio access networks: GERAN, UTRAN, and E-UTRAN. GERAN connects to an SGSN via Gb and A interfaces. UTRAN connects to an SGSN via Iu-ps and to an MSC via Iu-cs. E-UTRAN connects to an MME via S1-MME. The MME connects to the MSC via SGs and to the SGSN via S3. The SGSN connects to the MSC via Gs. The UE also has direct interfaces Um (to GERAN), Uu (to UTRAN), and LTE-Uu (to E-UTRAN).
+
+**Figure 1 – Network interfaces required for handling of CSFB procedures**
+
+Once CSFB has been successfully completed, both user and signalling plans corresponding to the current service session will be handled as in UTRAN or GERAN CS domains.
+
+Since this Recommendation focuses on CSFB impacts on voice service, the scope of discussion is limited to the aspects related to voice. It is also important to note that the transmission of SMS from/to UEs connected to E-UTRAN does not require a CSFB procedure and as such is not covered in this Recommendation.
+
+CSFB is of particular importance in all LTE networks where IP multimedia subsystem (IMS) core network or VoLTE is not available (either because they are not supported by the user equipment (UE) or by the network), since it is the only way UEs can access CS services, and in particular a critical one such as voice.
+
+Figures 1 to 3 (extracted from [ETSI TS 123 272]) represent the corresponding signalling flowcharts between these interfaces (respectively for a mobile originating context in Figure 2 and for a mobile terminating context in Figure 3).
+
+![Sequence diagram for Mobile originating call in active mode – No PS HO support. Lifelines: UE/ MS, eNodeB, BSS /RNS, MME, MSC, SGW/PGW, SGSN. The diagram shows the interaction between these components for a mobile originating call, including steps like NAS extended service request, S1-AP UE context modification, and CS MO call.](a5ee5c23b6dc52ec1d724b76d5a5f58f_img.jpg)
+
+```
+
+sequenceDiagram
+    participant UE/ MS
+    participant eNodeB
+    participant BSS /RNS
+    participant MME
+    participant MSC
+    participant SGW/PGW
+    participant SGSN
+
+    Note left of UE/ MS: 1a. NAS extended service request
+    UE/ MS->>MME: 1a. NAS extended service request
+    Note right of MME: 1d. MO CSFB indication
+    MME-->>MSC: 1d. MO CSFB indication
+    Note left of eNodeB: 1b. S1-AP UE context modification request with CS fallback indicator
+    MME->>eNodeB: 1b. S1-AP UE context modification request with CS fallback indicator
+    Note left of eNodeB: 1c. S1-AP UE context modification response message
+    eNodeB->>MME: 1c. S1-AP UE context modification response message
+
+    Note left of UE/ MS: 2. Optional measurement report solicitation
+    Note right of MME: 2. Optional measurement report solicitation
+
+    Note left of UE/ MS: 3a. NACC, 3b, 3c RRC connection release
+    eNodeB->>UE/ MS: 3a. NACC, 3b, 3c RRC connection release
+
+    Note left of eNodeB: 4. S1-AP: S1 UE context release request
+    eNodeB->>MME: 4. S1-AP: S1 UE context release request
+
+    Note left of UE/ MS: 5. S1 UE context release
+    Note right of MME: 5. S1 UE context release
+
+    Note left of UE/ MS: 6. UE changes RAT then LA update or RA update or LAU and RAU
+    Note right of MME: 6. UE changes RAT then LA update or RA update or LAU and RAU
+
+    Note left of UE/ MS: 7a. Suspend (see 23.060)
+    UE/ MS->>SGSN: 7a. Suspend (see 23.060)
+    Note left of MME: 7b. Suspend request/response
+    MME->>SGSN: 7b. Suspend request/response
+    Note left of MME: 8. Update bearer(s)
+    MME->>SGW/PGW: 8. Update bearer(s)
+
+    Note left of UE/ MS: 9. CM service request
+    UE/ MS->>BSS /RNS: 9. CM service request
+    Note left of MSC: 9. A/Iu-cs message (with CM service request)
+    BSS /RNS->>MSC: 9. A/Iu-cs message (with CM service request)
+
+    Note left of UE/ MS: 10a. Service reject
+    MSC->>UE/ MS: 10a. Service reject
+
+    Note left of UE/ MS: 10b. Location area update
+    Note right of MSC: If the MSC is changed
+    Note right of MME: 10b. Location area update
+    MSC->>MME: 10b. Location area update
+
+    Note left of UE/ MS: 10c. CS MO call
+    UE/ MS->>BSS /RNS: 10c. CS MO call
+    BSS /RNS->>MSC: 10c. CS MO call
+
+    Note left of UE/ MS: 11. Routing area update or combined RA/LA update
+    Note right of MME: 11. Routing area update or combined RA/LA update
+  
+```
+
+Sequence diagram for Mobile originating call in active mode – No PS HO support. Lifelines: UE/ MS, eNodeB, BSS /RNS, MME, MSC, SGW/PGW, SGSN. The diagram shows the interaction between these components for a mobile originating call, including steps like NAS extended service request, S1-AP UE context modification, and CS MO call.
+
+G.1028.2(19)\_F02
+
+**Figure 2 – Mobile originating call in active mode – No PS HO support [ETSI TS 123 272]**
+
+![Sequence diagram for Mobile terminating call in active mode - No PS HO support. Lifelines: UE/MS, eNodeB, BSS/RNS, MME, MSC, S-GW/PGW, SGSN. The diagram shows the interaction between these components for a CS service notification, paging, service request, and subsequent signaling connection release or CS connection establishment.](cfef993dcc8fb513de79eb1f93cf26ae_img.jpg)
+
+```
+
+sequenceDiagram
+    participant UE/MS
+    participant eNodeB
+    participant BSS/RNS
+    participant MME
+    participant MSC
+    participant S-GW/PGW
+    participant SGSN
+
+    Note left of UE/MS: 1A. CS SERVICE NOTIFICATION
+    UE/MS->>MME: 1b. NAS extended service request
+    MME->>MSC: 1A. PAGING REQUEST
+    MME->>MSC: 1A. SERVICE REQUEST
+    MSC-->>MME: 1c. CS paging reject
+    MME->>eNodeB: 1d. S1-AP UE context modification request with CS fallback indicator
+    eNodeB-->>MME: 1e. S1-AP UE context modification response message
+    Note left of UE/MS: 2. Optional measurement report
+    UE/MS->>BSS/RNS: 3a. CCO/NACC, 3b. 3c signalling connection release
+    eNodeB->>MME: 4. S1-AP: S1 UE CONTEXT RELEASE REQUEST
+    Note left of UE/MS: 5. S1 UE CONTEXT RELEASE
+    Note left of UE/MS: 6. UE changes RAT then, LAU OR RA UPDATE OR LAU AND RAU
+    UE/MS->>SGSN: 7a. Suspend (see TS 23.060)
+    MME->>SGSN: 7b. Suspend request/response
+    MME->>S-GW/PGW: 8. Update bearer(s)
+    UE/MS->>BSS/RNS: 9. Paging response
+    MSC-->>MME: 9a. Establish CS connection
+    Note right of MSC: IF THE MSC IS CHANGED
+    UE/MS->>BSS/RNS: 9b. Signalling connection release
+    BSS/RNS-->>MSC: 9b. CONNECTION REJECT
+    Note left of UE/MS: 9c. Location area update or combined RA/LA update
+  
+```
+
+Sequence diagram for Mobile terminating call in active mode - No PS HO support. Lifelines: UE/MS, eNodeB, BSS/RNS, MME, MSC, S-GW/PGW, SGSN. The diagram shows the interaction between these components for a CS service notification, paging, service request, and subsequent signaling connection release or CS connection establishment.
+
+G.1028.2(19)\_F03
+
+**Figure 3 – Mobile terminating call in active mode - No PS HO support [ETSI TS 123 272]**
+
+At the end of a CSFB call, the UE can re-register to the LTE network.
+
+This Recommendation focusses on the impact of the CSFB procedure on QoS and the assessment of this impact. Readers interested in a more detailed description and understanding of the procedure itself can refer to [ETSI TS 123 272] or to documents such as the ones provided in the bibliography.
+
+# 7 KPIs for circuit switched fallback
+
+End-to-end quality indicators provided in clause 9 (Table 2) of [ITU-T G.1028] apply for all calls starting under LTE coverage. This includes not only VoLTE calls, but also calls going into CSFB procedure. These metrics are:
+
+- Registration success rate
+- Service availability
+- Post dialing delay
+- Voice quality (MOS-LQ)
+- Mouth-to-ear delay
+- Call drop rate
+- Speech bandwidth (NB, WB or SWB)
+
+This clause provides further complementary indicators necessary to quantify the specific impact of this procedure on QoS.
+
+Moreover, this clause assumes that the UE is already connected to an LTE network and has therefore passed initial connection, authentication and attach phases.
+
+The relevant indicators from an end-to-end perspective, as perceived by an end-user located under a dual 4G and 3G/2G network coverage, are described in clauses 7.1 to 7.4, together with the potential contributor network key performance indicator(s) (KPI(s)).
+
+For all ITU-T G.1028 clause 9 (Table 2) end-to-end quality indicators, and for the set of indicators from an end-to-end perspective as perceived by an end-user described in clauses 7.1 to 7.4, the full context of the measurements should be supplied, including the time interval when the KPI measurements were collected, when reporting results.
+
+## 7.1 CSFB call setup failure ratio
+
+**Definition:** Number of calls successfully started by end users on a 4G/LTE network that are unsuccessfully switched to a 2G/3G network, divided by the total number of call attempts, expressed as a percentage.
+
+**Corresponding network KPIs and triggering points:** The CSFB procedure starts with the sending of the "RRC Connection Request" message with cause "mt\_Access" by the calling party or of the "EMM Extended Service request (CS fall back indicator)" message by called party. It ends successfully if the "Setup" Message is received from network.
+
+**Formula:**
+
+$$\text{CSFB Call Setup Failure Ratio [\%]} = \left( \frac{\text{unsuccessful CSFB call setup attempts}}{\text{Total number of CSFB call setup attempts}} \right) \times 100\%$$
+
+As defined in [ETSI TR 103 219], section 4.3.4.
+
+## 7.2 CSFB call set-up time
+
+**Definition:** The time it takes to perform the CSFB procedure, i.e., the total time required to start and complete the CSFB procedure from the UE of the calling party.
+
+From the end-user's perspective, this metric is strictly equivalent to post dialing delay as defined in ITU-T G.1028. It corresponds to the sum of the CSFB Time –calling party (as defined in [ETSI TR 103 219], section 4.3.3) and of the call setup time on the CS (2G or 3G) network.
+
+**Corresponding network KPIs and triggering points:** This metric corresponds to the measurement of the delay between the moment the "NAS CM Service Request" is sent with "CSFB requested" clause (or "RRC Connection Request" with cause "mo\_Data") to the reception of "NAS ALERTING" message on the target network.
+
+**Formula:**
+
+$$\text{CSFB Call Setup Time (s)} = (t_{\text{calling party receives alerting tone}} - t_{\text{user press call button}})$$
+
+## 7.3 CSFB return to 4G/LTE time
+
+**Definition:** The average time (in seconds) it takes for a UE to return to 4G from a legacy (2G/3G) network after a CSFB call has been ended by end-user.
+
+**Corresponding network KPIs and triggering points:** The return to 4G starts with the normal clearing message (NAS disconnect) and finishes when the first system information block (SIB) message has been received and decoded in LTE.
+
+### **Formula:**
+
+$$Return\ to\ 4G/LTE[s] = t_{\text{first System information block in LTE received}} - t_{\text{call disconnected}}$$
+
+As defined in [ETSI TR 103 219], section 4.3.7.
+
+NOTE – A meaningful measurement result may be the average of this metric.
+
+## 7.4 CSFB return to 4G/LTE failure ratio
+
+**Definition:** The probability that a UE is able, in a given time interval, to return to 4G from a legacy (2G/3G) network after a CSFB call has been ended by end-user.
+
+**Corresponding network KPIs and triggering points:** Similar trigger points to CSFB Return to 4G/LTE time, with in addition the possibility for the end trigger to be reached within a predefined time interval
+
+### **Formula:**
+
+$$Return\ to\ 4G\ LTE\ Failure\ Ratio\ (\%) = \left( 1 - \frac{\text{unsuccessful Return to LTE attempts}}{\text{Total Return to LTE attempts}} \right) * 100$$
+
+As defined in [ETSI TR 103 219], section 4.3.6.
+
+## 7.5 Quality targets
+
+Similar to the process in ITU-T G.1028 (Tables 4 to 7) for most of the indicators identified for VoLTE, a budget must be assigned for the indicators identified in clause 7 to the various segments that compose end-to-end paths, with indications of target values that can be reasonably reached on each of these segments.
+
+NOTE – Due to a lack of field feedback, this section is for further study. Table 1 will be completed once such feedback becomes available.
+
+**Table 1 – Quality budget allocation**
+
+| Network segment     | CSFB call setup failure ratio | CSFB call setup time | CSFB return to 4G/LTE time | CSFB return to 4G/LTE failure ratio |
+|---------------------|-------------------------------|----------------------|----------------------------|-------------------------------------|
+| UE                  |                               |                      |                            |                                     |
+| E-UTRAN             |                               |                      |                            |                                     |
+| UTRAN /GERAN        |                               |                      |                            |                                     |
+| EPC                 |                               |                      |                            |                                     |
+| IMS/AS              |                               |                      |                            |                                     |
+| <b>Total budget</b> |                               |                      |                            |                                     |
+
+# 8 Assessment methodology for CSFB
+
+The very nature of the CSFB procedure requires particular considerations when it comes to the measurement methods. In particular, since it involves correlation of events taking place in different radio access technologies (RATs), collection of information from different points and subsequent correlation is required in order to capture the requested information.
+
+However, a few similarities remain with the case of VoLTE discussed in [ITU-T G.1028], which are highlighted in the clauses that follows. The use cases for reporting, monitoring and troubleshooting described in ITU-T G.1028 also apply for CSFB use cases.
+
+NOTE – In complement to methods described below, crowdsourcing based approaches may be envisaged. Further study is, however, required in order to prove their practical and reliable application to collect technical KPIs for VoLTE and CSFB.
+
+## **8.1 Passive monitoring**
+
+Simultaneous monitoring of S1-Application Protocol (present in S1 interfaces, see measurement points C1 and D1 in ITU-T G.1028), SG and A/IuCS interfaces enable the capturing of the messages pertaining to the different phases of the CSFB procedures regardless of the initial state of the UE. Subsequent correlation (using unique identifiers such as IMSI, IMEI and activities timestamp) allows the connection of the different portions of the procedure and the generation of the required KPIs. This approach guarantees statistical validity of the samples since it will capture all CSFB attempts.
+
+## **8.2 Active monitoring**
+
+When using active monitoring (at either measurement points A2 or A3 as defined in ITU-T G.1028) it is important to identify the correct working conditions for the test set. Due to the very nature of this type of test, it can successfully be used to verify CSFB procedure in a restricted area (i.e., hospitals, airports, etc.) but cannot guarantee that the information collected can be representative of the situation on a larger scale. Specific biases may also be introduced by the type of device being used and its software release. As a result of these limitations it is recommended that active monitoring is used to monitor only specific locations and analysis is restricted on the trend over time of the collected data (i.e., improving/degrading) and not to use these data on a national scale to monitor networks performances.
+
+## **8.3 Drive testing**
+
+Considerations similar to those discussed for active monitoring in clause 8.2 can also be made for drive testing (see measurement point A3 in ITU-T G.1028). In particular, isolating the measurement conditions in place for such a complex procedure like CSFB (involving different passages in different networks) has a negative impact on the reliability of the measurements obtained using drive testing technologies.
+
+Similarly to what was previously discussed, parameters such as brand/model of the UE, OS software version as well as other aspects impacting radio frequency (RF) performances (e.g., utilization of internal or external antennas and their relative positioning when using multiple UEs at the same time) should be taken in account when using drive testing. Nevertheless, drive testing can provide useful insight on CSFB performance when considered on a punctual basis (i.e., using the obtained data as representative only for this specific moment in time/in space).
+
+## **8.4 Trace files/other network related information**
+
+If trace file functionality is enabled, it is possible to collect the information contained in these files related to CSFB event and to correlate them (using a post-processing tool) to obtain the specified KPIs. Specifications for the trace file formats and their contents vary from vendor to vendor and therefore specific analysis based on the technology to be monitored have to be put in place.
+
+# **9 Impact on QoS**
+
+This clause complements Tables A.1 and A.2 in Annex A of [ITU-T G.1028] with a list of degradations of QoS for voice services that are specific to CSFB.
+
+**Table 2 – Degradations related to call session performance and their potential causes**
+
+| <b>Degradation</b>           | <b>Possible reasons:</b>                                                                                                                                         | <b>Location</b>       |
+|------------------------------|------------------------------------------------------------------------------------------------------------------------------------------------------------------|-----------------------|
+| Unavailability of basic call | – Failure during any of the phases of CSFB procedure                                                                                                             | Terminal/eUTRAN/UTRAN |
+| Link failure                 | – Bad negotiation between two equipment of the network during CSFB procedure (bad codec management)                                                              | eUTRAN/UTRAN          |
+| PS session retention         | – In case of CSFB procedure, pre-existing data session on 4G may have to be handed over to 3G with a lower bit-rate, or suspended until the UE comes back to 4G. | Terminal/eUTRAN       |
+
+**Table 3 – Degradations related to perceived speech quality and their potential causes**
+
+| <b>Kind of degradation</b> | <b>Possible reasons:</b>                                                                                                   | <b>Location</b> |
+|----------------------------|----------------------------------------------------------------------------------------------------------------------------|-----------------|
+| Encoding/decoding issues   | – Narrowband instead of wideband speech quality<br>– No VoLTE available, CSFB procedure is applied back to NB CS telephony | Terminal/UTRAN  |
+
+# Bibliography
+
+- [b-Netmanias Tech-Blog]: Netmanias Tech-Blog, *Part-2: What happens when a user performs a voice call from an LTE/4G network? – VoLGA & CSFB*; accessed in August 2019.  
+<<https://www.netmanias.com/en/post/blog/10906/lte-volte/part-2-what-happens-when-a-user-performs-a-voice-call-from-an-lte-4g-network-volga-csfb>>
+
+
+
+## SERIES OF ITU-T RECOMMENDATIONS
+
+|                 |                                                                                                                                                           |
+|-----------------|-----------------------------------------------------------------------------------------------------------------------------------------------------------|
+| Series A        | Organization of the work of ITU-T                                                                                                                         |
+| Series D        | Tariff and accounting principles and international telecommunication/ICT economic and policy issues                                                       |
+| Series E        | Overall network operation, telephone service, service operation and human factors                                                                         |
+| Series F        | Non-telephone telecommunication services                                                                                                                  |
+| <b>Series G</b> | <b>Transmission systems and media, digital systems and networks</b>                                                                                       |
+| Series H        | Audiovisual and multimedia systems                                                                                                                        |
+| Series I        | Integrated services digital network                                                                                                                       |
+| Series J        | Cable networks and transmission of television, sound programme and other multimedia signals                                                               |
+| Series K        | Protection against interference                                                                                                                           |
+| Series L        | Environment and ICTs, climate change, e-waste, energy efficiency; construction, installation and protection of cables and other elements of outside plant |
+| Series M        | Telecommunication management, including TMN and network maintenance                                                                                       |
+| Series N        | Maintenance: international sound programme and television transmission circuits                                                                           |
+| Series O        | Specifications of measuring equipment                                                                                                                     |
+| Series P        | Telephone transmission quality, telephone installations, local line networks                                                                              |
+| Series Q        | Switching and signalling, and associated measurements and tests                                                                                           |
+| Series R        | Telegraph transmission                                                                                                                                    |
+| Series S        | Telegraph services terminal equipment                                                                                                                     |
+| Series T        | Terminals for telematic services                                                                                                                          |
+| Series U        | Telegraph switching                                                                                                                                       |
+| Series V        | Data communication over the telephone network                                                                                                             |
+| Series X        | Data networks, open system communications and security                                                                                                    |
+| Series Y        | Global information infrastructure, Internet protocol aspects, next-generation networks, Internet of Things and smart cities                               |
+| Series Z        | Languages and general software aspects for telecommunication systems                                                                                      |

marked/G/T-REC-G.107.2-202303-I_PDF-E/raw.md

@@ -0,0 +1,425 @@
+
+
+# Recommendation**ITU-T G.107.2 (03/2023)**
+
+SERIES G: Transmission systems and media, digital systems and networks
+
+International telephone connections and circuits –  
+Transmission planning and the E-model
+
+---
+
+## **Fullband E-model**
+
+![ITU logo](0538daaa5583c23e17db3a12f2281a55_img.jpg)
+
+The logo of the International Telecommunication Union (ITU) is located in the bottom right corner. It features a blue globe with white lines representing latitude and longitude, and the letters 'ITU' in a bold, blue, sans-serif font superimposed on the globe.
+
+ITU logo
+
+## ITU-T G-SERIES RECOMMENDATIONS **TRANSMISSION SYSTEMS AND MEDIA, DIGITAL SYSTEMS AND NETWORKS**
+
+|                                                                                                                                              |                    |
+|----------------------------------------------------------------------------------------------------------------------------------------------|--------------------|
+| INTERNATIONAL TELEPHONE CONNECTIONS AND CIRCUITS                                                                                             | G.100–G.199        |
+| <b>Transmission planning and the E-model</b>                                                                                                 | <b>G.100–G.109</b> |
+| General Recommendations on the transmission quality for an entire international telephone connection                                         | G.110–G.119        |
+| General characteristics of national systems forming part of international connections                                                        | G.120–G.129        |
+| General characteristics of the 4-wire chain formed by the international circuits and national extension circuits                             | G.130–G.139        |
+| General characteristics of the 4-wire chain of international circuits; international transit                                                 | G.140–G.149        |
+| General characteristics of international telephone circuits and national extension circuits                                                  | G.150–G.159        |
+| Apparatus associated with long-distance telephone circuits                                                                                   | G.160–G.169        |
+| Transmission plan aspects of special circuits and connections using the international telephone connection network                           | G.170–G.179        |
+| Protection and restoration of transmission systems                                                                                           | G.180–G.189        |
+| Software tools for transmission systems                                                                                                      | G.190–G.199        |
+| GENERAL CHARACTERISTICS COMMON TO ALL ANALOGUE CARRIER-TRANSMISSION SYSTEMS                                                                  | G.200–G.299        |
+| INDIVIDUAL CHARACTERISTICS OF INTERNATIONAL CARRIER TELEPHONE SYSTEMS ON METALLIC LINES                                                      | G.300–G.399        |
+| GENERAL CHARACTERISTICS OF INTERNATIONAL CARRIER TELEPHONE SYSTEMS ON RADIO-RELAY OR SATELLITE LINKS AND INTERCONNECTION WITH METALLIC LINES | G.400–G.449        |
+| COORDINATION OF RADIOTELEPHONY AND LINE TELEPHONY                                                                                            | G.450–G.499        |
+| TRANSMISSION MEDIA AND OPTICAL SYSTEMS CHARACTERISTICS                                                                                       | G.600–G.699        |
+| DIGITAL TERMINAL EQUIPMENTS                                                                                                                  | G.700–G.799        |
+| DIGITAL NETWORKS                                                                                                                             | G.800–G.899        |
+| DIGITAL SECTIONS AND DIGITAL LINE SYSTEM                                                                                                     | G.900–G.999        |
+| MULTIMEDIA QUALITY OF SERVICE AND PERFORMANCE – GENERIC AND USER-RELATED ASPECTS                                                             | G.1000–G.1999      |
+| TRANSMISSION MEDIA CHARACTERISTICS                                                                                                           | G.6000–G.6999      |
+| DATA OVER TRANSPORT – GENERIC ASPECTS                                                                                                        | G.7000–G.7999      |
+| PACKET OVER TRANSPORT ASPECTS                                                                                                                | G.8000–G.8999      |
+| ACCESS NETWORKS                                                                                                                              | G.9000–G.9999      |
+
+*For further details, please refer to the list of ITU-T Recommendations.*
+
+# Recommendation ITU-T G.107.2
+
+# Fullband E-model
+
+## Summary
+
+Recommendation ITU-T G.107.2 gives the algorithm for the fullband (FB) version of the E-model as the common ITU-T transmission rating model for planning speech services that provide FB speech transmission (20 – 20 000 Hz). This computational model can be useful to transmission planners, to help ensure that users will be satisfied with end-to-end transmission performance. The primary output of the model is a scalar rating of transmission quality. A major feature of this model is the use of transmission impairment factors that reflect the effects of different types of degradations occurring on the entire transmission path, mouth-to-ear.
+
+This FB-E-model is an adapted version of the narrowband (NB) (300 – 3 400 Hz) and wideband (WB) (50 – 7 000 Hz) E-models, which are described in Recommendations ITU-T G.107 NB and ITU-T G.107.1 WB. It does not replace the NB or the WB E-model. Instead, it describes a separate FB version of the model that uses, within limits, similar concepts and input parameters as the NB and WB E-models.
+
+## History
+
+| Edition | Recommendation | Approval   | Study Group | Unique ID*                                                                |
+|---------|----------------|------------|-------------|---------------------------------------------------------------------------|
+| 1.0     | ITU-T G.107.2  | 2019-06-29 | 12          | <a href="http://handle.itu.int/11.1002/1000/13926">11.1002/1000/13926</a> |
+| 2.0     | ITU-T G.107.2  | 2023-03-01 | 12          | <a href="http://handle.itu.int/11.1002/1000/15461">11.1002/1000/15461</a> |
+
+## Keywords
+
+E-model, fullband, speech quality, transmission planning.
+
+---
+
+\* To access the Recommendation, type the URL <http://handle.itu.int/> in the address field of your web browser, followed by the Recommendation's unique ID. For example, <http://handle.itu.int/11.1002/1000/11830-en>.
+
+## FOREWORD
+
+The International Telecommunication Union (ITU) is the United Nations specialized agency in the field of telecommunications, information and communication technologies (ICTs). The ITU Telecommunication Standardization Sector (ITU-T) is a permanent organ of ITU. ITU-T is responsible for studying technical, operating and tariff questions and issuing Recommendations on them with a view to standardizing telecommunications on a worldwide basis.
+
+The World Telecommunication Standardization Assembly (WTSA), which meets every four years, establishes the topics for study by the ITU-T study groups which, in turn, produce Recommendations on these topics.
+
+The approval of ITU-T Recommendations is covered by the procedure laid down in WTSA Resolution 1.
+
+In some areas of information technology which fall within ITU-T's purview, the necessary standards are prepared on a collaborative basis with ISO and IEC.
+
+## NOTE
+
+In this Recommendation, the expression "Administration" is used for conciseness to indicate both a telecommunication administration and a recognized operating agency.
+
+Compliance with this Recommendation is voluntary. However, the Recommendation may contain certain mandatory provisions (to ensure, e.g., interoperability or applicability) and compliance with the Recommendation is achieved when all of these mandatory provisions are met. The words "shall" or some other obligatory language such as "must" and the negative equivalents are used to express requirements. The use of such words does not suggest that compliance with the Recommendation is required of any party.
+
+## INTELLECTUAL PROPERTY RIGHTS
+
+ITU draws attention to the possibility that the practice or implementation of this Recommendation may involve the use of a claimed Intellectual Property Right. ITU takes no position concerning the evidence, validity or applicability of claimed Intellectual Property Rights, whether asserted by ITU members or others outside of the Recommendation development process.
+
+As of the date of approval of this Recommendation, ITU had not received notice of intellectual property, protected by patents/software copyrights, which may be required to implement this Recommendation. However, implementers are cautioned that this may not represent the latest information and are therefore strongly urged to consult the appropriate ITU-T databases available via the ITU-T website at <http://www.itu.int/ITU-T/ipr/>.
+
+© ITU 2023
+
+All rights reserved. No part of this publication may be reproduced, by any means whatsoever, without the prior written permission of ITU.
+
+## Table of Contents
+
+|                                                                       | Page |
+|-----------------------------------------------------------------------|------|
+| 1 Scope .....                                                         | 1    |
+| 2 References.....                                                     | 1    |
+| 3 Definitions .....                                                   | 2    |
+| 3.1 Terms defined elsewhere .....                                     | 2    |
+| 3.2 Terms defined in this Recommendation .....                        | 2    |
+| 4 Abbreviations and acronyms .....                                    | 2    |
+| 5 Conventions .....                                                   | 2    |
+| 6 Fullband E-model.....                                               | 2    |
+| 6.1 Introduction.....                                                 | 2    |
+| 6.2 Transmission rating scale of the FB E-model.....                  | 2    |
+| 7 The structure and basic algorithms of the fullband E-model.....     | 4    |
+| 7.1 Calculation of the transmission rating factor, R .....            | 4    |
+| 7.2 Basic signal-to-noise ratio, $R_{o,FB}$ .....                     | 4    |
+| 7.3 Simultaneous impairment factor, $I_{s,FB}$ .....                  | 5    |
+| 7.4 Delay impairment factor, $I_{d,FB}$ .....                         | 5    |
+| 7.5 Effective equipment impairment factor, $I_{e,eff,FB}$ .....       | 6    |
+| 7.6 Advantage factor, A .....                                         | 7    |
+| 7.7 Default values .....                                              | 7    |
+| Annex A MOS values derived from the transmission rating factor R..... | 8    |
+| Bibliography .....                                                    | 9    |
+
+
+
+# Recommendation ITU-T G.107.2
+
+## Fullband E-model
+
+# 1 Scope
+
+This Recommendation describes the fullband (FB) version of a computational model, known as the E-model, that has proven useful as a transmission planning tool for assessing the combined effects of variations in several transmission parameters that affect conversational<sup>1</sup> quality. This computational model can be used, for example, by transmission planners to help ensure that users will be satisfied with end-to-end transmission performance whilst avoiding over-engineering of networks. It must be emphasized that the primary output from the model is the "rating factor"  $R$ , but this can be transformed to give estimates of customer opinion. Such estimates are only made for transmission planning purposes and not for actual customer opinion prediction (for which there is no agreed-upon model recommended by the ITU-T).
+
+This version is an adapted version of the narrowband (NB) (300 – 3 400 Hz) and wideband (WB) (50 – 7 000 Hz) E-models, which are described in [ITU-T G.107] and [ITU-T G.107.1]. The FB version addresses scenarios which include FB (20 – 20 000 Hz) transmission. It does not replace the NB or the WB E-model. Instead, it describes a separate FB-version of the model that uses, within limits, similar concepts and input parameters as the NB and WB E-models. The current version only captures the effects of ambient background noise at the sending and receiving side, of the FB speech coding, voice-over-IP packet loss, and pure delay. Other degradations such as loudness loss, circuit noise, talker echo, listener echo, non-optimum sidetone levels and quantizing distortions have not yet been covered by the model.
+
+Regarding the interpretation of the FB E-model ratings, note that the current versions of [b-ITU-T G.108], [b-ITU-T G.108.1], and [b-ITU-T G.109] do not refer to the FB version described here, but only to the NB version of the E-model described in [ITU-T G.107].
+
+# 2 References
+
+The following ITU-T Recommendations and other references contain provisions which, through reference in this text, constitute provisions of this Recommendation. At the time of publication, the editions indicated were valid. All Recommendations and other references are subject to revision; users of this Recommendation are therefore encouraged to investigate the possibility of applying the most recent edition of the Recommendations and other references listed below. A list of the currently valid ITU-T Recommendations is regularly published. The reference to a document within this Recommendation does not give it, as a stand-alone document, the status of a Recommendation.
+
+[ITU-T G.107] Recommendation ITU-T G.107 (2015), *The E-model: a computational model for use in transmission planning*.  
+<<http://www.itu.int/rec/T-REC-G.107>>
+
+[ITU-T G.107.1] Recommendation ITU-T G.107.1 (2019), *Wideband E-model*.  
+<<https://www.itu.int/rec/T-REC-G.107.1/en>>
+
+---
+
+<sup>1</sup> Conversational quality in this context refers to transmission characteristics, e.g., long transmission times, effects of talker echoes, etc. However, the E-model, as described in this Recommendation, has not yet been developed to an extent which would cover those characteristics in a realistic way. Most of its formulae, except the ones given in clause 7.4, have been developed on the basis of subjective listening-only experiments. The model is also not intended to model transmission impairments during double talk situations. These effects are for further study within ITU-T Study Group 12.
+
+- [ITU-T G.113] Recommendation ITU-T G.113 (2007), *Transmission impairments due to speech processing*.  
+<<http://www.itu.int/rec/T-REC-G.113>>
+- [ITU-T G.722] Recommendation ITU-T G.722 (2012), *7 kHz audio-coding within 64 kbit/s*.  
+<<http://www.itu.int/rec/T-REC-G.722>>
+- [ITU-T P.800] Recommendation ITU-T P.800 (1996), *Methods for subjective determination of transmission quality*.  
+<<http://www.itu.int/rec/T-REC-P.800>>
+
+# **3 Definitions**
+
+## **3.1 Terms defined elsewhere**
+
+None.
+
+## **3.2 Terms defined in this Recommendation**
+
+None.
+
+# **4 Abbreviations and acronyms**
+
+This Recommendation uses the following abbreviations and acronyms:
+
+- ACR Absolute Category Rating
+- FB Fullband
+- MOS Mean Opinion Score
+- NB Narrowband
+- SWB Super-wideband
+- WB Wideband
+
+# **5 Conventions**
+
+None.
+
+# **6 Fullband E-model**
+
+## **6.1 Introduction**
+
+The complexity of modern networks requires that, for transmission planning, the many transmission parameters be not only considered individually, but also that their combined effects be taken into account. This can be done by "expert, informed guessing", but a more systematic approach is desirable, such as using a computational model. The output from the model described here is a scalar transmission rating value,  $R$ , which varies directly with the overall conversational quality. [ITU-T G.113] gives guidance about specific impairments, including the combined effects based upon a simplification of the model.
+
+## **6.2 Transmission rating scale of the FB E-model**
+
+For the narrowband (NB) case described in [ITU-T G.107], the transmission rating scale ranges from  $R = 0$  (lowest possible quality) to  $R = 100$  (optimum quality). On this scale, a default NB transmission channel including logarithmic PCM coding and a noise floor (default parameter values according to Table 2 of [ITU-T G.107]) obtains a rating of  $R = 93.2$ . For a wideband (WB) speech transmission channel, the quality is generally judged better than that for an NB channel. Thus, this scale range was extended in [ITU-T G.107.1] to a maximum value of  $R = 129$  for a clean wideband (50 – 7 000 Hz)
+
+channel, as it is defined in [ITU-T G.722]. In the present version, this scale was further extended to reflect the even higher quality of the fullband (20 – 20 000 Hz) channel.
+
+Unfortunately, it is not possible to obtain direct human judgements on the *R*-scale, as this scale has additivity properties that are not reflected by the ordinary rating scales. Instead, for NB conditions, ITU-T recommends collecting judgements on a 5-point absolute category rating (ACR) scale, see [ITU-T P.800]. The mean rating, averaged over all the test participants and stimuli reflecting the same circuit condition, is then called a mean opinion score (MOS).
+
+It has been shown that the MOS ratings differ between tests where only NB stimuli are presented, and tests where both NB / WB or purely WB stimuli are presented, since the use of the scale is largely influenced by the stimulus set. On the other hand, there is also experimental evidence that judgements for WB samples collected in a purely WB context do not differ significantly from those collected in a mixed NB / WB context (see [b-Barriac] and [b-Takahashi]). In addition to the stimulus bandwidth, test results are influenced by the test participant group, the language, the participants' native country, etc. [b-Möller]. For an NB context, an average S-shaped relationship is defined between the *R*-scale (range [0; 100]) and MOS ratings (range [1; 4.5]) collected from the "average" test participants in an "average" experimental setting, see Annex B and Appendix I of [ITU-T G.107].
+
+For a WB or a mixed NB / WB context, the *R*-scale is extended in a way that leaves the NB use of the scale unaffected, including the position of the reference connection (default parameter settings according to Table 2 of [ITU-T G.107]). The extension is based on pairs of auditory tests in which the same (NB) test stimuli have been judged once in a purely NB, and once in a mixed NB / WB context. The judgements on these common stimuli define a relationship between the use of the MOS-scale in an NB and in a mixed NB / WB context [b-Raake].
+
+The same principle is applied for extending the *R*-scale beyond the wideband case. First, mixed bandwidth databases that contain a direct super-wideband (SWB) (20 – 16 000 Hz) condition, as well as NB / WB codecs for which the WB equipment impairment factor (*I<sub>e</sub>, WB*) is known, are identified. The subjective MOS values of the mixed-band conditions have been transformed to the *R*-scale with the transformations given in [ITU-T G.107] and [ITU-T G.107.1] which are valid for the NB and WB case. *R<sub>WB/SWB</sub>* then presents the *R* values of the NB / WB / SWB mixed bandwidth database compressed onto the WB *R*-scale of [ITU-T G.107.1] (which is limited to 129), whereas the *R<sub>WB</sub>* presents their uncompressed *R* values. A simple linear regression is used to map the compressed *R<sub>WB/SWB</sub>* values (derived from the subjective MOS) onto *R<sub>WB</sub>* (derived from the equipment impairment factors). The linear regression is calculated without an intercept term and therefore the estimation model goes through the origin. The *R<sub>WB</sub>* value which corresponds to *R<sub>WB/SWB</sub>* = 129 then corresponds to the maximum value *R<sub>max</sub>* of the *R*-scale in the SWB case. Figure 1 shows an exemplary extrapolation for one database.
+
+![Figure 1: Comparison between R-values derived in an NB / WB and in a mixed NB / WB / SWB context. The graph plots R_WB on the y-axis (0 to 200) against R_WB/SWB on the x-axis (0 to 140). A diagonal line represents the 1:1 relationship. Data points are shown as red circles, with two specific points labeled: R, wb/swb = 129 and R, wb = 149.2. The label G.107.2(19)_F01 is present in the bottom right corner of the plot area.](adb1f42239329fa8283d1a40005f989f_img.jpg)
+
+| $R_{WB/SWB}$ | $R_{WB}$ |
+|--------------|----------|
+| ~55          | ~65      |
+| ~70          | ~75      |
+| ~75          | ~85      |
+| ~95          | ~110     |
+| ~100         | ~120     |
+| 129          | 149.2    |
+
+Figure 1: Comparison between R-values derived in an NB / WB and in a mixed NB / WB / SWB context. The graph plots R\_WB on the y-axis (0 to 200) against R\_WB/SWB on the x-axis (0 to 140). A diagonal line represents the 1:1 relationship. Data points are shown as red circles, with two specific points labeled: R, wb/swb = 129 and R, wb = 149.2. The label G.107.2(19)\_F01 is present in the bottom right corner of the plot area.
+
+**Figure 1 – Comparison between  $R$ -values derived in an NB / WB and in a mixed NB / WB / SWB context**
+
+Comparing the maximum  $R_{max}$  values obtained with different databases, it is decided that a value of  $R_{max} = 148$  should be used as the maximum for a super-wideband case. As experimental evidence suggests that super-wideband (SWB) and fullband (FB) conditions are perceptually equivalent when being evaluated on a 5-point ACR scale according to [ITU-T P.800], it is decided to also use a value of  $R_{max} = 148$  in the fullband case.
+
+It should be noted that the primary output of the FB E-model is the transmission rating  $R$ . However, the output can also give nominal estimates of user reactions, for instance in the form of MOS values, as described in Annex A.
+
+# 7 The structure and basic algorithms of the fullband E-model
+
+The FB E-model is based on the equipment impairment factor method, following previous transmission rating models. The model estimates the conversational quality from mouth-to-ear as perceived by the user at the receive side, both as listener and talker.
+
+The current version of the FB E-model only considers the effects of the background noise at the sending and receiving side, of coding-decoding (codec) degradations, the effects of packet loss and discard in voice-over-IP telephony, as well as the effects of pure delay. It assumes that headsets with diotic listening are used on both sides of the connection.
+
+## 7.1 Calculation of the transmission rating factor, $R$
+
+For FB, the basic E-model formula (7-1 of [ITU-T G.107]) can be re-written as:
+
+$$R = R_{o,FB} - I_{s,FB} - I_{d,FB} - I_{e,eff,FB} + A \quad (7-1)$$
+
+$R_{o,FB}$  represents in principle the basic signal-to-noise ratio. The factor  $I_{s,FB}$  is a combination of all impairments which occur more or less simultaneously with the voice signal. The factor  $I_{d,FB}$  represents the impairments occurring delayed with respect to the speech signal. The effective equipment impairment factor  $I_{e,eff,FB}$  represents impairments caused by low bit-rate codecs. It also includes impairment due to randomly distributed or bursty pack losses. The advantage factor  $A$  allows for compensation of impairment factors when the user benefits from other types of access. The following clauses give the equations used in the FB E-model.
+
+## 7.2 Basic signal-to-noise ratio, $R_{o,FB}$
+
+The basic signal-to-noise ratio  $R_{o,FB}$  is defined by:
+
+$$Ro,FB = 20 - 1.5 \cdot (SLR + No,FB) \quad (7-2)$$
+
+The term  $No,FB$  (in dBm0p) is the power addition of different noise sources:
+
+$$No,FB = 10 \cdot \log \left[ 10^{\frac{Nc}{10}} + 10^{\frac{Nos,FB}{10}} + 10^{\frac{Nor,FB}{10}} + 10^{\frac{Nfo}{10}} \right] \quad (7-3)$$
+
+$Nc$  represents the psophometrically weighted power of the circuit noise, and  $Nfo$  the electrical power of noise on the receiving line referred to as the 0 dBr point. Both are set to  $-96$  dBm0p.
+
+$Nos,FB$  (in [dBm0p]) is the equivalent circuit noise at the 0 dBr point, caused by the A-weighted room noise  $Ps$  at the send side:
+
+$$Nos,FB = Ps - SLR - Ds - 100 + 0.004 \cdot (Ps - OLR - Ds - 14)^2 \quad (7-4)$$
+
+where  $OLR = SLR + RLR$ . In the same way, the room noise  $Pr$  at the receive side is transferred into an equivalent circuit noise  $Nor$  (in [dBm0p]) at the 0 dBr point.
+
+$$Nor,FB = RLR - 147 + 1.12 \cdot Pre + 0.009 \cdot (Pre - 25)^2 \quad (7-5)$$
+
+The term  $Pre$  (in [dBm0p]) is the "effective room noise" caused by the enhancement of  $Pr$  by the listener's sidetone path:
+
+$$Pre = Pr + 10 \cdot \log_{10} \left( 1 + 10^{\frac{10-LSTR}{10}} \right) \quad (7-6)$$
+
+## 7.3 Simultaneous impairment factor, $Is,FB$
+
+The factor  $Is,FB$  is the sum of all impairments which may occur more or less simultaneously with the voice transmission. This aspect has not been analysed for the FB case so far, thus it is set to:
+
+$$Is,FB = 0 \quad (7-7)$$
+
+## 7.4 Delay impairment factor, $Id,FB$
+
+The factor  $Id,FB$ , represents all impairments due to the delay of voice signals.
+
+So far, only the contribution of pure delay, i.e., without echo, is addressed by the FB E-model, which results in  $Id,FB = Idd$ . Only when the effect due to pure delay is attributed to the service, is the effect, to some extent, reflected in the respective speech quality evaluation by the user. It is noted that the effect due to pure delay may instead be attributed to the other conversation partner, or may be attributed to a conversational difficulty. For example, if the conversation partners do not know each other, the delay may simply reduce the efficiency of the communication. In the current version of the E-model, different types of conversations and/or users are considered, in terms of their interactivity and delay-sensitivity. As a consequence, the predictions can be tailored to the delay sensitivity requirements of the user group.
+
+The contribution for the pure delay is thus calculated from the overall one-way delay  $Ta$ , as well as the minimum perceivable delay  $mT$ , and the delay sensitivity  $sT$  as follows:
+
+For  $Ta \leq mT$ :
+
+$$Idd = 0 \quad (7-8a)$$
+
+For  $Ta > mT$ :
+
+$$Idd = 1.48 \cdot 25 \left\{ \left( 1 + X^{6 \cdot sT} \right)^{\frac{1}{6 \cdot sT}} - 3 \left( 1 + \left[ \frac{X}{3} \right]^{6 \cdot sT} \right)^{\frac{1}{6 \cdot sT}} + 2 \right\} \quad (7-8b)$$
+
+with:
+
+$$X = \frac{\log\left(\frac{Ta}{mT}\right)}{\log 2} \quad (7-9)$$
+
+The settings of delay sensitivity,  $sT$ , and minimum perceivable delay,  $mT$ , as indicated in Table 1 are recommended, reflecting different use cases and user groups. Two aspects are addressed by these settings:
+
+- The interactivity of the conversation and the sensitivity of the users to the delay-effect.
+- The application scenario, that is, whether a given call is being made in a business context or in an everyday situation. Even if users may not notice the delay, it may be very critical for the efficiency or even effectiveness of a given call, for example in a business context.
+
+As a consequence, in the case where it is uncertain what user group or what application scenario is being addressed with the planned service, it is recommended that the default class is used. Any case of non-default value usage should be explicitly mentioned when reporting results.
+
+Based on these considerations, the following settings shown in Table 1 are recommended for  $sT$  and  $mT$ :
+
+**Table 1 – Delay-sensitivity classes for different use cases**
+
+| Class of delay-sensitivity | $sT$ | $mT$ (ms) | Use case                                                                                                                                                                                                                                                                                                                                                                            |
+|----------------------------|------|-----------|-------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------|
+| Default                    | 1    | 100       | <ul style="list-style-type: none"> <li>– Applicable to all types of telephone conversations.</li> <li>– Must be used for:               <ul style="list-style-type: none"> <li>• Carrier-grade fixed or mobile telephony;</li> <li>• Enterprise-grade fixed or mobile telephony;</li> <li>• When the targeted user group and delay requirements are unknown.</li> </ul> </li> </ul> |
+| Medium                     | 0.55 | 120       | <ul style="list-style-type: none"> <li>– Applicable only in cases where it is known that users have an average sensitivity to delay, e.g., conversations similar to random number verification scenarios.</li> </ul>                                                                                                                                                                |
+| Low                        | 0.4  | 150       | <ul style="list-style-type: none"> <li>– Applicable only in cases where it is known that users have very low sensitivity to delay, e.g., Conversations similar to short conversation test scenarios.</li> </ul>                                                                                                                                                                     |
+
+## 7.5 Effective equipment impairment factor, $Ie,eff,FB$
+
+The effective equipment impairment factor  $Ie,eff,FB$  is derived using the codec-specific value for the equipment impairment factor at zero packet-loss  $Ie,FB$ , the packet-loss robustness factor  $Bpl$ , and the burstiness robustness factor  $Brf$ . With the packet-loss probability  $Ppl$  and the burst ratio  $BurstR$ ,  $Ie,eff,FB$  is calculated using the equation:
+
+$$Ie,eff,FB = Ie,FB + (132 - Ie,FB) \cdot \frac{Ppl - \frac{1-BurstR}{Brf}}{Ppl+Bpl} \quad (7-10)$$
+
+$BurstR$  is the so-called burst ratio, which is defined as
+
+$$BurstR = \frac{\text{Average length of observed bursts in an arrival sequence}}{\text{Average length of observed bursts for the network under "random" loss}} \quad (7-11)$$
+
+When packet loss is random (i.e., independent)  $BurstR = 1$ ; and when packet loss is bursty (i.e., dependent)  $BurstR > 1$ . For calculations with no packet loss (i.e.,  $Ppl = 0$ ) the burstiness is defined as randomly distributed ( $BurstR = 1$ ).
+
+For example, for packet loss distributions corresponding to a 2-state Markov model with transition probabilities  $p$  between a "found" and a "loss" state, and  $q$  between the "loss" and the "found" state, the burst ratio can be calculated as:
+
+$$BurstR = \frac{1}{p+q} = \frac{Ppl \div 100}{p} = \frac{1 - (Ppl \div 100)}{q} \quad (7-12)$$
+
+The burstiness robustness factor *Brf* is a codec-specific value that reflects the codec robustness to the burstiness of pack loss. A higher burstiness robustness factor indicates an increased robustness against the packet loss of the codec under study. If the burstiness robustness of a codec is unknown, the *Brf* should be set to 1.
+
+As seen from equation 7-10, the effective equipment impairment factor in the case of *Ppl* = 0 (no packet-loss) is equal to the *Ie,FB* value.
+
+Corresponding values for *Ie,FB*, *Bpl* and *Brf* can be found in Appendix V of [ITU-T G.113].
+
+## 7.6 Advantage factor, *A*
+
+Background information on the advantage factor *A* can be found in Appendix II of [ITU-T G.113]. As this effect has not yet been studied for the FB case, it is recommended to set:
+
+$$A = 0 \quad (7-13)$$
+
+## 7.7 Default values
+
+For all input parameters used in the algorithm of the E-model, the default values are listed in Table 2. It is strongly recommended to use these default values for all the parameters which are not varied during the planning calculation.
+
+**Table 2 – Default values and permitted ranges for the parameters**
+
+| Parameter                                | Abbr.         | Unit  | Default value | Permitted range | Remark       |
+|------------------------------------------|---------------|-------|---------------|-----------------|--------------|
+| Send loudness rating                     | <i>SLR</i>    | dB    | +8            | 0 ... +18       | (Note 1)     |
+| Receive loudness rating                  | <i>RLR</i>    | dB    | +2            | –5 ... +14      | (Note 1)     |
+| Sidetone masking rating                  | <i>STMR</i>   | dB    | 15            | 10 ... 20       | (Note 2)     |
+| Listener sidetone rating                 | <i>LSTR</i>   | dB    | 18            | 13 ... 23       | (Note 2)     |
+| D-value of telephone, send side          | <i>Ds</i>     | –     | 3             | –3 ... +3       | (Note 2)     |
+| D-value of telephone, receive side       | <i>Dr</i>     | –     | 3             | –3 ... +3       | (Note 2)     |
+| Circuit noise referred to as 0 dBr-point | <i>Nc</i>     | dBm0p | –96           | –               | (Note 1)     |
+| Noise floor at the receive side          | <i>Nfo</i>    | dBm0p | –96           | –               | (Note 1)     |
+| Equipment impairment factor              | <i>Ie,FB</i>  | –     | 0             | 0 ... 120       | (Note 3)     |
+| Packet-loss robustness factor            | <i>Bpl</i>    | –     | 4.3           | 7.4 ... 18      | (Notes 1, 3) |
+| Random packet-loss probability           | <i>Ppl</i>    | %     | 0             | 0 ... 20        | (Notes 1, 3) |
+| Burst ratio                              | <i>BurstR</i> | –     | 1             | 1 ... 8         | (Note 1)     |
+| Burstiness robustness factor             | <i>Brf</i>    | –     | 1             | 1 ... 20        | (Notes 1, 3) |
+| Overall one-way delay                    | <i>Ta</i>     | ms    | 0             | 0 ... 1 700     |              |
+| Delay sensitivity                        | <i>sT</i>     | –     | 1             | 0.4 ... 1       | (Note 1)     |
+| Minimum perceivable delay                | <i>mT</i>     | ms    | 100           | 20 ... 200      | (Note 1)     |
+| Advantage factor                         | <i>A</i>      | –     | 0             | 0 ... 20        |              |
+
+NOTE 1 – Currently under study.  
+NOTE 2 – Fixed relation: *LSTR* = *STMR* + *D*.  
+NOTE 3 – If *Ppl* > 0%, then the *Bpl* and *Brf* must match the codec, packet size, and PLC assumed.
+
+## Annex A
+
+### MOS values derived from the transmission rating factor $R$
+
+(This annex forms an integral part of this Recommendation.)
+
+The transmission rating factor  $R$  can be in the range from 0 to 148, where  $R = 0$  represents an extremely bad quality, and  $R = 148$  represents a very high quality in the FB case. An estimated mean opinion score ( $MOS_{CQEF}$ ) for the conversational situation on a scale of 1-5 can be obtained from the  $R$ -factor by using the equations:
+
+$$Rx = \frac{R}{1.48} \quad (A-1)$$
+
+For  $Rx < 0$ :  $MOS_{CQEF} = 1$
+
+For  $0 < Rx < 100$ :  $MOS_{CQEF} = 1 + 0.035 Rx + Rx (Rx - 60)(100 - Rx) \cdot 7 \cdot 10^{-6}$  (A-2)
+
+For  $Rx > 100$ :  $MOS_{CQEF} = 4.5$
+
+## Bibliography
+
+- [b-ITU-T G.108] Recommendation ITU-T G.108 (1999), *Application of the E-model: A planning guide*.
+- [b-ITU-T G.108.1] Recommendation ITU-T G.108.1 (2000), *Guidance for assessing conversational speech transmission quality effects not covered by the E-Model*.
+- [b-ITU-T G.109] Recommendation ITU-T G.109 (1999), *Definition of categories of speech transmission quality*.
+- [b-Barriac] Barriac, V., Le Saout, J.-Y., and Lockwood C. (2004), *Discussion on unified methodologies for the comparison of voice quality of narrowband and wideband scenarios*.  
+<[https://www.google.com/url?sa=t&rct=j&q=&esrc=s&source=web&cd=&cad=rja&uact=8&ved=2ahUKEWjk3u79kbf-AhXaumMGHccxDEUQFnoECAwQAQ&url=https%3A%2F%2Fportal.etsi.org%2Fstq%2Fworks%2Fhop2004%2Fpresentations2004%2F090-mainz\\_ft\\_objective\\_methodologies.doc&usg=AOvVaw0](https://www.google.com/url?sa=t&rct=j&q=&esrc=s&source=web&cd=&cad=rja&uact=8&ved=2ahUKEWjk3u79kbf-AhXaumMGHccxDEUQFnoECAwQAQ&url=https%3A%2F%2Fportal.etsi.org%2Fstq%2Fworks%2Fhop2004%2Fpresentations2004%2F090-mainz_ft_objective_methodologies.doc&usg=AOvVaw0)>
+- [b-Möller] Möller, S. (2000), *Assessment and Prediction of Speech Quality in Telecommunications*, Springer.  
+<<https://link.springer.com/book/10.1007/978-1-4757-3117-0>>
+- [b-Raake] Raake, A. (2006), *Speech Quality of VoIP – Assessment and Prediction*, Wiley.  
+<<https://www.wiley.com/en-ie/Speech+Quality+of+VoIP:+Assessment+and+Prediction-p-9780470030608>>
+- [b-Takahashi] Takahashi, A., Kurashima, A., and Yoshino, H. (2005), *Subjective Quality Index for Compatibly Evaluating Narrowband and Wideband Speech*, MESAQIN.  
+<<https://cir.nii.ac.jp/crid/1570009750682169472>>
+
+
+
+
+
+## SERIES OF ITU-T RECOMMENDATIONS
+
+|                 |                                                                                                                                                           |
+|-----------------|-----------------------------------------------------------------------------------------------------------------------------------------------------------|
+| Series A        | Organization of the work of ITU-T                                                                                                                         |
+| Series D        | Tariff and accounting principles and international telecommunication/ICT economic and policy issues                                                       |
+| Series E        | Overall network operation, telephone service, service operation and human factors                                                                         |
+| Series F        | Non-telephone telecommunication services                                                                                                                  |
+| <b>Series G</b> | <b>Transmission systems and media, digital systems and networks</b>                                                                                       |
+| Series H        | Audiovisual and multimedia systems                                                                                                                        |
+| Series I        | Integrated services digital network                                                                                                                       |
+| Series J        | Cable networks and transmission of television, sound programme and other multimedia signals                                                               |
+| Series K        | Protection against interference                                                                                                                           |
+| Series L        | Environment and ICTs, climate change, e-waste, energy efficiency; construction, installation and protection of cables and other elements of outside plant |
+| Series M        | Telecommunication management, including TMN and network maintenance                                                                                       |
+| Series N        | Maintenance: international sound programme and television transmission circuits                                                                           |
+| Series O        | Specifications of measuring equipment                                                                                                                     |
+| Series P        | Telephone transmission quality, telephone installations, local line networks                                                                              |
+| Series Q        | Switching and signalling, and associated measurements and tests                                                                                           |
+| Series R        | Telegraph transmission                                                                                                                                    |
+| Series S        | Telegraph services terminal equipment                                                                                                                     |
+| Series T        | Terminals for telematic services                                                                                                                          |
+| Series U        | Telegraph switching                                                                                                                                       |
+| Series V        | Data communication over the telephone network                                                                                                             |
+| Series X        | Data networks, open system communications and security                                                                                                    |
+| Series Y        | Global information infrastructure, Internet protocol aspects, next-generation networks, Internet of Things and smart cities                               |
+| Series Z        | Languages and general software aspects for telecommunication systems                                                                                      |

marked/G/T-REC-G.126-199303-I_PDF-E/raw.md

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+
+
+![ITU logo: A globe with a lightning bolt and the letters ITU.](2dfa6ac3edfe874f68aa0cbccaa42322_img.jpg)
+
+ITU logo: A globe with a lightning bolt and the letters ITU.
+
+INTERNATIONAL TELECOMMUNICATION UNION
+
+**ITU-T**
+
+**G.126**
+
+TELECOMMUNICATION  
+STANDARDIZATION SECTOR  
+OF ITU
+
+(03/93)
+
+**TRANSMISSION SYSTEMS AND MEDIA**
+
+**GENERAL CHARACTERISTICS OF NATIONAL  
+SYSTEMS FORMING PART OF INTERNATIONAL  
+CONNECTIONS**
+
+---
+
+**LISTENER ECHO IN TELEPHONE NETWORKS**
+
+**ITU-T Recommendation G.126**
+
+(Previously "CCITT Recommendation")
+
+---
+
+# FOREWORD
+
+The ITU Telecommunication Standardization Sector (ITU-T) is a permanent organ of the International Telecommunication Union. The ITU-T is responsible for studying technical, operating and tariff questions and issuing Recommendations on them with a view to standardizing telecommunications on a worldwide basis.
+
+The World Telecommunication Standardization Conference (WTSC), which meets every four years, established the topics for study by the ITU-T Study Groups which, in their turn, produce Recommendations on these topics.
+
+ITU-T Recommendation G.126 was revised by the ITU-T Study Group XII (1988-1993) and was approved by the WTSC (Helsinki, March 1-12, 1993).
+
+---
+
+# NOTES
+
+1 As a consequence of a reform process within the International Telecommunication Union (ITU), the CCITT ceased to exist as of 28 February 1993. In its place, the ITU Telecommunication Standardization Sector (ITU-T) was created as of 1 March 1993. Similarly, in this reform process, the CCIR and the IFRB have been replaced by the Radiocommunication Sector.
+
+In order not to delay publication of this Recommendation, no change has been made in the text to references containing the acronyms "CCITT, CCIR or IFRB" or their associated entities such as Plenary Assembly, Secretariat, etc. Future editions of this Recommendation will contain the proper terminology related to the new ITU structure.
+
+2 In this Recommendation, the expression "Administration" is used for conciseness to indicate both a telecommunication administration and a recognized operating agency.
+
+# CONTENTS
+
+|                                                                              | <i>Page</i> |
+|------------------------------------------------------------------------------|-------------|
+| 1 Introduction and scope .....                                               | 1           |
+| 2 Definitions.....                                                           | 1           |
+| 3 Effect of listener echo .....                                              | 3           |
+| 3.1 General.....                                                             | 3           |
+| 3.2 Effect on voice transmission.....                                        | 3           |
+| 3.3 Effect on voice-band data transmission .....                             | 4           |
+| 3.4 Effect of listener echo in a connection with multiple 4-wire loops ..... | 4           |
+| 4 Limits of listener echo .....                                              | 5           |
+| 4.1 Voice transmission.....                                                  | 5           |
+| 4.2 Voice-band data transmission.....                                        | 5           |
+| References .....                                                             | 7           |
+
+
+
+# LISTENER ECHO IN TELEPHONE NETWORKS
+
+(Helsinki, 1993)
+
+# 1 Introduction and scope
+
+A telephone connection between two subscribers in telephone networks usually contains one or more 2-4-2-wire loops (further referred to as 4-wire loop), as the connection may encounter a mixture of 2- and 4-wire analogue and digital switching and transmission systems, including PBXs. Signal reflections may occur due to the impedance mismatch at both hybrids that terminate a 4-wire loop. A portion of the main signal will thus be reflected at the far end of the 4-wire loop, return to the near end and be reflected again. Therefore the reflected signal will finally arrive at the listener's end some time later than the original signal. The delayed signal is called listener echo.
+
+Listener echo signals can:
+
+- lead to objectionable "hollowness" in voice transmissions; and
+- impair the bit error ratio of voice-band data transmissions.
+
+The effect of listener echo on transmission performance can be characterized by the additional loss and additional delay in the listener echo path relative to the direct signal path. The minimum value of the additional listener echo path loss over the frequency band involved provides a margin against singing distortion. Recommendation G.122 provides guidance on the influence of national networks on stability and talker echo in international connections. This Recommendation provides planning guidance for listener echo in telephone networks with the aim of obtaining satisfactory speech and voice-band data transmission performance.
+
+# 2 Definitions
+
+For the purpose of this Recommendation, the following definitions apply:
+
+**2.1 listener echo (receive end echo):** echo produced by double reflected signals and disturbing the listener, receiving voice-band data equipment, etc.
+
+# NOTES
+
+1 The term "received end echo" is a term preferred by some Administrations.
+
+2 With small delay against the wanted signal (less than about 3 ms) listener echo may cause hollowness in telephony. In transmission of voice-band data signals, listener echo may cause bit errors and, in any case, reduces the margin against other disturbances.
+
+**2.2 listener echo loss (receive echo loss):** degree of attenuation of the double reflected signal with respect to the wanted signal. In terms of the absolute losses of both signals, the listener echo loss is (see Figure 1/G.100):  $LE = L_2 - L_1$ .
+
+NOTE – For practical purposes the listener echo loss is equal to the open-loop loss (valid if the latter exceeds 8 dB). The listener echo loss characterizes the degree of disturbance by hollowness, as well as the disturbing effect on voice-band data modem receivers.
+
+**2.3 hollowness:** distortion in telephony caused by double reflected signals and subjectively perceived as a "hollow sound", i.e. as if the talker would speak into some hollow vessel.
+
+NOTE – Hollowness is to be distinguished from listener echo.
+
+**2.4 open-loop loss (OLL):** in a loop formed by a 4-wire circuit (or a cascade connection of two or more 4-wire circuits) and terminated by 2-wire ends (i.e. having “4-wire terminating sets”, or hybrids, at both ends), the loss measured by breaking the loop at some point, injecting a signal and measuring the loss incurred in traversing the open loop. All impedance conditions should be preserved while making the measurement. See Figure 2/G.100.
+
+#### NOTES
+
+- 1 In practice the OLL is equal to the listener echo loss.
+- 2 The OLL is also equal to the sum of the two semi-loop losses associated with a loop.
+
+**2.5 path a-t-b (transmission loss of ...); semi-loop loss:** the transmission loss between the points “a” and “b” of the 4-wire termination (as defined at the virtual switching points) independent of whether there exists or not a physical point “t”.
+
+### **2.5.1 Possible alternative to the definition in 2.5**
+
+#### **Semi-loop loss**
+
+In an arrangement comprising a 4-wire circuit (or a cascade connection of several 4-wire circuits) with unwanted coupling between the go and return direction at the ends of the circuit - usually via a 4-wire terminating set, or via acoustical coupling - the loss measured between the input and output. See Figure 3/G.100.
+
+## NOTES
+
+- 1 The semi-loop loss is an important quantity in determining echo balance return loss, echo loss, listener echo loss (see also open-loop loss).
+- 2 Distinction may be made between the semi-loop loss of a given piece of equipment and the semi-loop loss of a national system. The latter is measured at equi-level points in an ISC which serves as a national gateway exchange.
+
+**2.6 singing margin (SM):** the minimum listener echo loss in dB over the frequency band involved.
+
+**2.7 round-trip delay (DL):** the delay in ms around the closed 4-wire loop, determined primarily by the two-way delay of the 4-wire transmission path, which is equal to listener echo path delay.
+
+**2.8 weighted listener echo path loss (WEPL):** WEPL is the weighted mean value of listener echo loss expressed by the following equation:
+
+$$WEPL = -20 \log_{10} \frac{1}{3200} \int_{200}^{3400} 10^{-\frac{EPL(f)}{20}} df$$
+
+Where
+
+EPL (*f*) = magnitude of listener echo loss in dB at the frequency *f*.
+
+This concept was originally used in North America, in the transmission rating model, which can be used to derive the subjectively equivalent effects of listener echo on voice transmission performance regardless of the frequency response of the listener echo loss in the connection.
+
+# 3 Effect of listener echo
+
+## 3.1 General
+
+Figure 1 shows a connection with one 4-wire loop. The loss of the direct signal path is assumed to be  $S$  dB, whereas the loss along the path of the double-reflected signal is  $L$  dB. The listener echo loss (LE) is then  $(L-S)$  dB. It can be seen from Figure 1 that if signals are reflected twice only, the listener echo loss LE is equal to the loss around the 4-wire loop (Open-Loop Loss, OLL), as the other path losses are incurred equally by the direct and the reflected signals. Obviously, the additional delay of signals reflected twice is also equal to that around the 4-wire loop (round trip delay, DL), primarily determined by the two-way delay of the 4-wire transmission path.
+
+![Diagram of a 4-wire loop showing signal paths and losses.](997233d405f0d4b89ddeb7683e047f66_img.jpg)
+
+A schematic diagram of a 4-wire loop. A central rectangular box represents the 4-wire loop. Two input lines enter from the left, and two output lines exit to the right. On the left, a square box represents a transceiver, with a curved arrow indicating a signal path from the input line to the transceiver and back to the input line. On the right, another square box represents a second transceiver, also with a curved arrow indicating a signal path. Above the central box, a straight line with an arrow pointing right represents the direct signal path, labeled 'S'. Below the central box, a curved line with an arrow pointing left represents the double-reflected signal path, labeled 'L'. The text 'T1203540-91/d01' is located in the bottom right corner of the diagram.
+
+Diagram of a 4-wire loop showing signal paths and losses.
+
+FIGURE 1/G.126
+
+### Listener echo
+
+The loss around the 4-wire loop can be measured by breaking the loop at some point, injecting a signal from one end and measuring the loss incurred in traversing the open loop at the other end. All impedance conditions of the closed loop at the 2-wire ends should be preserved while making the measurement. The measured quantity is the Open Loop Loss (OLL), which is also equal to the sum of the two semi-loop losses (SLL) associated with the 4-wire loop. For practical purposes, semi-loop measurements may be more easily carried out, especially in the case of 4-wire exchanges with 2-wire circuit terminations, since it is sometimes difficult to maintain a connection through a 4-wire exchange and interrupt one direction of transmission. The definition of SLL and its measurement are illustrated in Figure 3/G.100.
+
+## 3.2 Effect on voice transmission
+
+The effect of listener echo on voice transmission performance will be dependent on the open loop loss and round-trip delay of 4-wire loops. It should be noted that usually the listener echo will consist of a series of signals reflected two times, four times, etc. and hence LE and OLL are in principle not equal. In practice however, LE and OLL may be taken as equal when OLL exceeds 8 dB.
+
+The results of some experiments show that the effect of listener echo on voice transmission performance will be negligible when OLL is greater than 10 dB with the round-trip delay less than 10 ms [1].
+
+Considering that the listener echo loss (or OLL) is normally shaped with frequencies and its minimum value (Singing Margin, SM) often occurs at the edge of the voice-frequency band, the minimum OLL can be taken as a margin against instability but does not provide the optimal measure for assessing the subjective quality of a near-singing connection [2]. The experiment results reported in Supplement No. 3 of CCITT *Blue Book* Volume V suggest that the weighted value of OLL over the voice-frequency band is more appropriate for evaluating the effect of listener echo on voice transmission performance. Based on the transmission rating model described in Supplement No. 3 of Volume V, the transmission quality index may be used to evaluate the effect of listener echo combined with other impairment factors on subjective speech transmission performance in connections.
+
+## 3.3 Effect on voice-band data transmission
+
+Listener echo causes ripples in the frequency response of the received voice-band data signal and impairs its bit error ratio. In order to obtain satisfactory transmission performance, the value of OLL in the band 500 to 2500 Hz should be higher than that for voice transmission in the band 300 - 3400 Hz due to the strict performance requirement of voice-band data transmission. In addition, the round-trip delay factor may be disregarded when characterizing the effect of listener echo on voice-band data transmission performance, since voice-band data error performance does not appear to be related to the delay. The results of some experiments show that the OLL required for satisfactory voice-band data transmission mainly depends on the data transmission speed, but sometimes also on the type of modems [3].
+
+## 3.4 Effect of listener echo in a connection with multiple 4-wire loops
+
+If there is more than one 4-wire loop in a connection, several reflected signals will accumulate at the receiving end as shown in Figure 2 (for the sake of simplicity, higher order reflections are not shown).
+
+![Diagram illustrating signal reflections in a connection with multiple 4-wire loops. A 'Talker' is on the left, and a 'Listener' is on the right. Three '4-wire loop' blocks are connected in series. Arrows show the forward signal path and multiple reflected paths (dashed lines) that accumulate at the listener end. The diagram is labeled 'T1203550-91/d02'.](35a7554182eb055209552843f341a1ae_img.jpg)
+
+The diagram illustrates the signal path from a 'Talker' on the left to a 'Listener' on the right through three sequential 4-wire loops. Each loop is depicted as a square containing a circle with internal arrows showing signal circulation. A solid horizontal line represents the primary signal path moving from left to right. Dashed lines represent reflected signals that loop back within each 4-wire loop and then proceed forward, accumulating at the listener's end. The diagram is labeled 'T1203550-91/d02' in the bottom right corner.
+
+Diagram illustrating signal reflections in a connection with multiple 4-wire loops. A 'Talker' is on the left, and a 'Listener' is on the right. Three '4-wire loop' blocks are connected in series. Arrows show the forward signal path and multiple reflected paths (dashed lines) that accumulate at the listener end. The diagram is labeled 'T1203550-91/d02'.
+
+FIGURE 2/G.126  
+**Signal reflections in a connection with multiple 4-wire loops**
+
+In a connection with multiple 4-wire loops, the effect of listener echo on transmission performance depends not only on the OLL of each 4-wire loop, but also on the one-way loss of each 4-wire loop and the 2-wire circuit loss between adjacent 4-wire loops. The one-way loss of 4-wire loops and the 2-wire circuit loss govern the interaction among 4-wire loops [4].
+
+Because of accumulated impairments and interactions, the OLL of each 4-wire loop in a connection with multiple 4-wire loops has to be much higher than that in a single 4-wire loop connection in order to achieve the same transmission performance. In a connection with multiple 4-wire loops, the effect of listener echo on voice-band data transmission is extremely significant. References [4] and [5] illustrate the relationship between voltage and power accumulation of listener echoes in a connection with multiple 4-wire loops. In general, uncorrelated echoes should be assumed for multiple 4-wire loop connections and the power addition criterion should be used to achieve satisfactory performance. This means that the OLL required for each 4-wire loop should be increased by  $10 \log m$  dB, where *m* is the total number of 4-wire loops. However, for connections with very low loss, the required increment for the OLL of each 4-wire loop may be  $16.6 \log m$  dB and for connections with intermediate loss, the addition of  $13.3 \log m$  dB may be necessary.
+
+# 4 Limits of listener echo
+
+## 4.1 Voice transmission
+
+The degree to which listener echo affects the quality of a voice connection depends on two factors:
+
+- 1) the Weighted Listener Echo Path Loss (WEPL); and
+- 2) the round trip delay.
+
+Transmission quality degrades when the WEPL is reduced and when the round trip delay is increased. See 2.8 for the definition of WEPL. The transmission planner must select appropriate values of WEPL and round trip delay to achieve a desired level of transmission quality for connections in a target population. The target population is that set of connections which it is, possible to construct and which involves the listener echo path or paths of interest. For example, take the simple case of connections involving subscriber lines connected through a digital local office. If any of the  $N$  subscriber lines can connect to any of the other  $N$  lines, then the target population of connections consists of the  $N(N-1)/2$  possibilities.
+
+Once the transmission planner has identified the target population of connections and the desired level of transmission quality, then the information contained in Recommendation P.11 and Supplement No. 3 can be used to calculate the minimum value of WEPL and maximum value of round trip delay required to provide that degree of transmission quality. To illustrate the procedure, an example is given in Figure 3 and Table 1. The information in the figure and table was derived for use in the North American network. Using the North American Transmission Rating Model found in Supplement No. 3, values of WEPL and round trip delay were calculated such that listener echo would add only a small additional impairment to the overall transmission quality. The objective that was used was a reduction of no more than 2% Good or Better average grade of service. The target population was the connections of subscriber loops through digital local offices.
+
+The curves in Figure 3 and data in Table 1 are to be interpreted as follows. For a given desired (or required) value of round trip delay, no more than 50% of the connections in the target population may have WEPL less than the value given by the curve marked "50", no more than 5% of connections may have WEPL less than the value given by the curve marked "95", and so forth. For example, if the round trip delay is specified to be 4 ms, then in the target population no more than 50% of connections may have  $WEPL \leq 18$  dB, no more than 5% may have  $WEPL \leq 13$  dB, and no more than 1% may have  $WEPL \leq 9$  dB. In addition to the WEPL, the minimum value of singing margin for any connection in the target population must conform to Recommendation G.122.
+
+After identifying the required maximum round trip delay, minimum WEPL, and minimum singing margin for the target population of connections, the transmission planner must then make sure that the equipment in the network [e.g. subscriber lines and telephones, 2/4-wire hybrids and balancing method(s)] and the network transmission loss/level plan are adequate to assure that these requirements are met for the target population of connections. This is not always easy to do. For example in the North American network it was necessary to equip the digital local switches with adjustable echo balancing networks, which can be automatically adjusted to achieve the best echo balance with the mix of subscriber lines in the network.
+
+## 4.2 Voice-band data transmission
+
+The following consideration provides an example and can serve as an indication of what values of OLL might be required by existing types of modems with a bit rate of up to 2.4 kbit/s, in order to obtain high quality data transmission:
+
+- a complete connection should not contain more than five (exceptionally seven) physical loops;
+- loops with very high OLL (exceeding, e.g. 45 dB) need not be included in the number of loops in the connection;
+- the OLL of each loop at any frequency in the band 500-2500 Hz should not be less than the values indicated in Table 2 (based on  $OLL = 18 + 10 \log m$ , where  $m$  = total number of loops).
+
+![Graph showing Weighted Listener Echo Path Loss (WEPL) objectives in dB versus Round-trip delay in ms. The graph includes four curves for 50%, 95%, 99%, and 99.9% of connections. The y-axis (WEPL) ranges from 5 to 30 dB, and the x-axis (Round-trip delay) ranges from 0 to 30 ms. The curves show that as the round-trip delay increases, the required WEPL also increases. Higher percentiles of connections require higher WEPL values for the same delay.](adb1f42239329fa8283d1a40005f989f_img.jpg)
+
+| Round-trip delay (ms) | WEPL (dB) for 50% connections | WEPL (dB) for 99.9% connections |
+|-----------------------|-------------------------------|---------------------------------|
+| 4                     | 18                            | 9                               |
+| 6                     | 20                            | 11                              |
+| 8                     | 22                            | 12                              |
+
+Graph showing Weighted Listener Echo Path Loss (WEPL) objectives in dB versus Round-trip delay in ms. The graph includes four curves for 50%, 95%, 99%, and 99.9% of connections. The y-axis (WEPL) ranges from 5 to 30 dB, and the x-axis (Round-trip delay) ranges from 0 to 30 ms. The curves show that as the round-trip delay increases, the required WEPL also increases. Higher percentiles of connections require higher WEPL values for the same delay.
+
+T1203560-91/d03
+
+FIGURE 3/G.126  
+**Weighted Listener Echo Path Loss (WEPL) objectives**
+
+TABLE 1/G.126
+
+**Planned values of WEPL**
+
+| Round-trip delay (ms) | Minimum WEPL (dB)        |                     |
+|-----------------------|--------------------------|---------------------|
+|                       | For 99.9% of connections | For 50% connections |
+| 4                     | 9                        | 18                  |
+| 6                     | 11                       | 20                  |
+| 8                     | 12                       | 22                  |
+
+NOTE – For connections with multiple 4-wire loops an increment of  $10 \log m$  dB for WEPL of each 4-wire loop may be required, where  $m$  is the number of loops. The values is under study.
+
+TABLE 2/G.126
+
+### **OLL and number of loops in a connection**
+
+| In one national system   |                  | Maximum total number of loops in international connection |
+|--------------------------|------------------|-----------------------------------------------------------|
+| Number of national loops | OLL of each loop |                                                           |
+| 1                        | 22,8 dB          | 3                                                         |
+| 2                        | 25 dB            | 5                                                         |
+| 3                        | 26,5 dB          | 7                                                         |
+
+NOTE – In general, the OLL in a single 4-wire loop connection may need to be at least 25 dB for high speed data with a bit rate of 4.8 kbit/s up to 9.8 kbit/s at any frequency in the band 500-2500 Hz.
+
+# **References**
+
+- [1] CCITT Contribution COM XII-No. 208, study period 1985-1988.
+- [2] CAVANAUGH (J.R.), HATCH (R.W.) and NEIGH (J.L.): A model for the subjective effects of listener echo on telephone connections, *B.S.T.J.*, Vol. 59, No. 6, pp. 1009-1060, July-August 1980.
+- [3] CCITT Contribution COM XII-No. 173, study period 1985-1988.
+- [4] GRUBER (J.G.): Performance analysis and design guidelines for multiple 4-wire path telephone connections, *IEEE Trans. on Communications*, Vol. COM 34, pp. 647-653, July 1986.
+- [5] CCITT Contribution COM XII-No. 42, study period 1981-1984.

marked/G/T-REC-G.142-199812-I_PDF-E/raw.md

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+
+
+![ITU logo](2dfa6ac3edfe874f68aa0cbccaa42322_img.jpg)
+
+The logo of the International Telecommunication Union (ITU) features the letters 'ITU' in a bold, sans-serif font, superimposed on a stylized globe with intersecting lines.
+
+ITU logo
+
+INTERNATIONAL TELECOMMUNICATION UNION
+
+**ITU-T**
+
+TELECOMMUNICATION  
+STANDARDIZATION SECTOR  
+OF ITU
+
+**G.142**
+
+(12/98)
+
+SERIES G: TRANSMISSION SYSTEMS AND MEDIA,  
+DIGITAL SYSTEMS AND NETWORKS
+
+International telephone connections and circuits – General  
+characteristics of the 4-wire chain of international circuits;  
+international transit
+
+# --- **Transmission characteristics of exchanges**
+
+ITU-T Recommendation G.142
+
+(Previously CCITT Recommendations)
+
+---
+
+## ITU-T G-SERIES RECOMMENDATIONS **TRANSMISSION SYSTEMS AND MEDIA, DIGITAL SYSTEMS AND NETWORKS**
+
+|                                                                                                                                              |                    |
+|----------------------------------------------------------------------------------------------------------------------------------------------|--------------------|
+| INTERNATIONAL TELEPHONE CONNECTIONS AND CIRCUITS                                                                                             | G.100–G.199        |
+| General definitions                                                                                                                          | G.100–G.109        |
+| General Recommendations on the transmission quality for an entire international telephone connection                                         | G.110–G.119        |
+| General characteristics of national systems forming part of international connections                                                        | G.120–G.129        |
+| General characteristics of the 4-wire chain formed by the international circuits and national extension circuits                             | G.130–G.139        |
+| <b>General characteristics of the 4-wire chain of international circuits; international transit</b>                                          | <b>G.140–G.149</b> |
+| General characteristics of international telephone circuits and national extension circuits                                                  | G.150–G.159        |
+| Apparatus associated with long-distance telephone circuits                                                                                   | G.160–G.169        |
+| Transmission plan aspects of special circuits and connections using the international telephone connection network                           | G.170–G.179        |
+| Protection and restoration of transmission systems                                                                                           | G.180–G.189        |
+| Software tools for transmission systems                                                                                                      | G.190–G.199        |
+| <b>INTERNATIONAL ANALOGUE CARRIER SYSTEM</b>                                                                                                 |                    |
+| GENERAL CHARACTERISTICS COMMON TO ALL ANALOGUE CARRIER-TRANSMISSION SYSTEMS                                                                  | G.200–G.299        |
+| INDIVIDUAL CHARACTERISTICS OF INTERNATIONAL CARRIER TELEPHONE SYSTEMS ON METALLIC LINES                                                      | G.300–G.399        |
+| GENERAL CHARACTERISTICS OF INTERNATIONAL CARRIER TELEPHONE SYSTEMS ON RADIO-RELAY OR SATELLITE LINKS AND INTERCONNECTION WITH METALLIC LINES | G.400–G.449        |
+| COORDINATION OF RADIOTELEPHONY AND LINE TELEPHONY                                                                                            | G.450–G.499        |
+| <b>TESTING EQUIPMENTS</b>                                                                                                                    |                    |
+| <b>TRANSMISSION MEDIA CHARACTERISTICS</b>                                                                                                    | G.600–G.699        |
+| <b>DIGITAL TRANSMISSION SYSTEMS</b>                                                                                                          |                    |
+| TERMINAL EQUIPMENTS                                                                                                                          | G.700–G.799        |
+| DIGITAL NETWORKS                                                                                                                             | G.800–G.899        |
+| DIGITAL SECTIONS AND DIGITAL LINE SYSTEM                                                                                                     | G.900–G.999        |
+
+*For further details, please refer to ITU-T List of Recommendations.*
+
+# **TRANSMISSION CHARACTERISTICS OF EXCHANGES**
+
+## **Summary**
+
+This Recommendation was revised to remove text that referred to international analogue exchanges. The ITU-T no longer recommends the placement of new or continued operation of analogue international exchanges. Additionally, this Recommendation was written so as to recognize that both Plesiochronous Digital Hierarchy/Synchronous Digital Hierarchy (PDH/SDH) and digital ATM switching technology may be used in the PSTN. This Recommendation provides transmission performance planning guidance for digital exchanges.
+
+###### **Source**
+
+ITU-T Recommendation G.142 was revised by ITU-T Study Group 12 (1997-2000) and was approved under the WTSC Resolution No. 1 procedure on the 3<sup>rd</sup> of December 1998.
+
+## FOREWORD
+
+ITU (International Telecommunication Union) is the United Nations Specialized Agency in the field of telecommunications. The ITU Telecommunication Standardization Sector (ITU-T) is a permanent organ of the ITU. The ITU-T is responsible for studying technical, operating and tariff questions and issuing Recommendations on them with a view to standardizing telecommunications on a worldwide basis.
+
+The World Telecommunication Standardization Conference (WTSC), which meets every four years, establishes the topics for study by the ITU-T Study Groups which, in their turn, produce Recommendations on these topics.
+
+The approval of Recommendations by the Members of the ITU-T is covered by the procedure laid down in WTSC Resolution No. 1.
+
+In some areas of information technology which fall within ITU-T's purview, the necessary standards are prepared on a collaborative basis with ISO and IEC.
+
+## NOTE
+
+In this Recommendation the term *recognized operating agency (ROA)* includes any individual, company, corporation or governmental organization that operates a public correspondence service. The terms *Administration*, *ROA* and *public correspondence* are defined in the *Constitution of the ITU (Geneva, 1992)*.
+
+## INTELLECTUAL PROPERTY RIGHTS
+
+The ITU draws attention to the possibility that the practice or implementation of this Recommendation may involve the use of a claimed Intellectual Property Right. The ITU takes no position concerning the evidence, validity or applicability of claimed Intellectual Property Rights, whether asserted by ITU members or others outside of the Recommendation development process.
+
+As of the date of approval of this Recommendation, the ITU had not received notice of intellectual property, protected by patents, which may be required to implement this Recommendation. However, implementors are cautioned that this may not represent the latest information and are therefore strongly urged to consult the TSB patent database.
+
+© ITU 1999
+
+All rights reserved. No part of this publication may be reproduced or utilized in any form or by any means, electronic or mechanical, including photocopying and microfilm, without permission in writing from the ITU.
+
+## CONTENTS
+
+###### Page
+
+|       |                                                   |   |
+|-------|---------------------------------------------------|---|
+| 1     | Scope .....                                       | 1 |
+| 2     | General .....                                     | 1 |
+| 3     | References .....                                  | 1 |
+| 4     | Digital exchanges .....                           | 2 |
+| 4.1   | Digital processes – Effect on transmission.....   | 2 |
+| 4.2   | Transmission loss through a digital exchange..... | 4 |
+| 4.3   | Relative levels .....                             | 5 |
+| 4.4   | Echo and stability control.....                   | 5 |
+| 4.4.1 | Transmission loss contribution .....              | 5 |
+| 4.4.2 | Balance return loss contribution.....             | 6 |
+| 4.5   | Local transmission.....                           | 7 |
+| 4.6   | Sidetone and input impedance.....                 | 7 |
+| 4.7   | Digital pads.....                                 | 8 |
+| 4.8   | Transmission delay.....                           | 8 |
+
+
+
+# **TRANSMISSION CHARACTERISTICS OF EXCHANGES**
+
+*(Geneva, 1980; amended at Melbourne, 1988, revised in 1998)*
+
+## **1 Scope**
+
+This Recommendation is concerned with the voice-frequency transmission considerations that should be taken into account in the design of digital exchanges and their incorporation into the network. The digital exchanges referred to include local exchanges and transit exchanges (national and international). The transmission considerations relate primarily to the properties which digital exchanges should possess to enable them to operate under different and changing network conditions with respect to the content of analogue, mixed analogue/digital and all-digital plant.
+
+## **2 General**
+
+Detailed transmission characteristics for PDH/SDH digital exchanges are contained in Recommendations Q.551, Q.552, Q.553 and Q.554. While no detailed Recommendations have been specifically generated for switches which use ATM technology, in general, ATM switches should comply with the requirements defined in the Q.550 series Recommendations. It is understood that due to the different mode of operation of ATM switches, as compared to their PDH/SDH counterparts, certain parameters like group delay may be different for an ATM switch that performs cell assembly when information from one DS-0 is placed within a specific cell.
+
+## **3 References**
+
+The following ITU-T Recommendations and other references contain provisions which, through reference in this text, constitute provisions of this Recommendation. At the time of publication, the editions indicated were valid. All Recommendations and other references are subject to revision; all users of this Recommendation are therefore encouraged to investigate the possibility of applying the most recent edition of the Recommendations and other references listed below. A list of the currently valid ITU-T Recommendations is regularly published.
+
+- ITU-T Recommendation G.101 (1996), *The transmission plan*.
+- ITU-T Recommendation G.113 (1996), *Transmission impairments*.
+- ITU-T Recommendation G.121 (1993), *Loudness Ratings (LRs) of national systems*.
+- ITU-T Recommendation G.122 (1993), *Influence of national systems on stability and talker echo in international connections*.
+- ITU-T Recommendation G.126 (1993), *Listener echo in telephone networks*.
+- ITU-T Recommendation G.176 (1997), *Planning guidelines for the integration of ATM technology into networks supporting voiceband services*.
+- ITU-T Recommendation Q.551 (1996), *Transmission characteristics of digital exchanges*.
+- ITU-T Recommendation Q.552 (1996), *Transmission characteristics at 2-wire analogue interfaces of digital exchanges*.
+
+- ITU-T Recommendation Q.553 (1996), *Transmission characteristics at 4-wire analogue interfaces of digital exchanges*.
+- ITU-T Recommendation Q.554 (1996), *Transmission characteristics at digital interfaces of digital exchanges*.
+- ITU-T Supplement 31 to the G-series Recommendations (1993), *Principles of determining an impedance strategy for the local network*.
+
+## 4 Digital exchanges
+
+### 4.1 Digital processes – Effect on transmission
+
+Digital PDH/SDH exchanges and digital ATM exchanges, to varying degrees, currently are required to include such digital processes as analogue-to-digital coders, digital-to-analogue decoders and digital recording processes, examples of which are companding law converters and digital pads. The extent to which such digital processes might be included in a digital exchange is determined by the network environment in which the exchange is to operate (i.e. mixed analogue/digital or all-digital). While a number of signal processing functions such as echo control, speech compression and PDH/SDH to ATM interworking may be performed by physically distinct network elements, this does not preclude the incorporation of these digital signal processing functions into digital switches.
+
+Digital processes such as those referred to above attract transmission penalties. These penalties can be expressed in terms of "quantization distortion units (qdus) or units of equipment impairment factor (eif)". The impact of these impairments on a connection can be evaluated using the "quantization factor method" or the "equipment impairment factor method", as discussed in Recommendation G.113. Only the equipment impairment factor method which is discussed in clause 6/G.113 is recommended for evaluating the impairments associated with transmission devices which use non-waveform coding techniques.
+
+A limit is placed on the permissible accumulation of units of transmission impairment in an international telephone connection. Details of the planning rule resulting from this limit and the penalties introduced by individual digital processes are given in 5.6/G.113.
+
+In accordance with 5.6/G.113, it is recommended that no more than 14 units of transmission impairment be permitted to accumulate in an international connection. Of these 14 units, a maximum of 5 units could be introduced by each national extension and a maximum of 4 units by the international portion<sup>1</sup>. Since one 8-bit PCM codec pair (coder and decoder) introduces 1 unit of transmission impairment, it is clear that unintegrated PCM digital processes involving analogue/digital conversions (e.g. codecs), or digital processes involving the recoding of information (e.g. digital pads) should not be allowed to proliferate in an uncontrolled fashion. Figure 1 shows some of the transmission paths that might be established through a digital exchange and the "units of transmission impairment" attributable to the digital processes in these paths. Figure 2 also shows some of the transmission paths that might be established through a digital exchange. However, Figure 2 is intended to identify examples of where digital signal processing devices may be placed. It is recommended that the impairments caused by these signal processing devices be evaluated using the equipment impairment factor method presented in clause 6/G.113. It is important to understand that the equipment impairment factor method allows the performance evaluation to be performed if the impairments are specified in impairment units of eifs and or qdus.
+
+<sup>1</sup> It should be noted that 14 qdus equates to approximately 20 eif units.
+
+![Diagram of transmission paths at digital exchanges showing seven cases (Case 1 to Case 7) with various components like digital exchange, transmission, switching, pads, and terminating units. Includes a legend for symbols like Analogue transmission, Digital transmission, A/D coder or D/A decoder, Analogue pad, Digital pad, 2W/4W terminating unit, Balance network, 0.7* qd units of transmission impairment, R Receive pad, T Transmit pad, L Pad for circuit loss adjustment, and Digital switch.](ebff22fb5dd6f50a90e44dca0f82f285_img.jpg)
+
+The diagram illustrates seven transmission cases (Case 1 to Case 7) at a digital exchange. The exchange is divided into 'Transmission' and 'Switching' sections. Each case shows a different configuration of analogue and digital transmission paths, including various pads (L, T, R) and terminating units (B). The impairment values are indicated in units of 0.7\* qd.
+
+- Case 1:** Pure digital transmission path through the exchange to a digital switch.
+- Case 2:** Digital transmission path with digital pads (L) of 0.7\* impairment in the switching section before the digital switch.
+- Case 3:** Analogue transmission path entering via a 2W/4W terminating unit (B) with transmit (T) and receive (R) analogue pads of 1/2\* impairment.
+- Case 4:** Analogue transmission path with a 2W/4W terminating unit (B) and T/R pads (1/2\* impairment), plus digital pads (L) of 0.7\* impairment before the digital switch.
+- Case 5:** Analogue transmission path with an analogue pad (L) of 1/2\* impairment in the switching section.
+- Case 6:** Analogue transmission path with an analogue pad (L) of 1/2\* impairment and a digital pad (L) of 0.7\* impairment.
+- Case 7:** Analogue transmission path with an analogue pad (L) of 1/2\* impairment located in the transmission section before the digital exchange boundary.
+
+|                                                                                                                                                                                                |                                                                                                                                                                                                                                                                              |
+|------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------|------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------|
+| <p>—— Analogue transmission</p> <p>– – – Digital transmission</p> <p>— — A/D coder or D/A decoder</p> <p>—I— Analogue pad</p> <p>– -I- – Digital pad</p> <p>—  B  — 2W/4W terminating unit</p> | <p>B Balance network</p> <p>0.7* qd units of transmission impairment</p> <p>R Receive (from the switch) pad at local exchange</p> <p>T Transmit (from the switch) pad at local exchange</p> <p>L Pad (if required) for circuit loss adjustment</p> <p>–X– Digital switch</p> |
+|------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------|------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------|
+
+T1208970-98
+
+Diagram of transmission paths at digital exchanges showing seven cases (Case 1 to Case 7) with various components like digital exchange, transmission, switching, pads, and terminating units. Includes a legend for symbols like Analogue transmission, Digital transmission, A/D coder or D/A decoder, Analogue pad, Digital pad, 2W/4W terminating unit, Balance network, 0.7\* qd units of transmission impairment, R Receive pad, T Transmit pad, L Pad for circuit loss adjustment, and Digital switch.
+
+**Figure 1/G.142 – Transmission paths at digital exchanges**
+
+![Diagram of transmission paths at digital exchanges (Cases 8-14) and a legend for symbols and components.](a5ee5c23b6dc52ec1d724b76d5a5f58f_img.jpg)
+
+The diagram illustrates various transmission paths at a digital exchange, categorized into 'Transmission' and 'Switching' sections. Cases 8 through 13 show different combinations of Half Echo Cancellers (HEC), Digital Circuit Multiplication Equipment (DCME), and ATM Interworking Functions (IWF) connected to digital switches. Case 14 shows an analogue transmission path through a balance network (B) and an ATM IWF, with transmit (T) and receive (R) pads at the local exchange, each having a 1/2 qd unit of transmission impairment. A legend at the bottom defines the symbols and components used.
+
+**Legend:**
+
+- Analogue transmission
+- - - - Digital transmission
+- |— A/D coder or D/A decoder
+- |— Analogue pad
+- |— Digital pad
+- |— 2W/4W terminating unit
+- B Balance network
+- 0.7\* qd units of transmission impairment
+- R Receive (from the switch) pad at local exchange
+- T Transmit (from the switch) pad at local exchange
+- P Pad (if required) for circuit loss adjustment
+- X— Digital switch
+- ATM IWF ATM to PDH/SDH Interworking Function
+- DCME Digital Circuit Multiplication Equipment
+- HEC Half Echo Canceller
+
+T1208960-98
+
+Diagram of transmission paths at digital exchanges (Cases 8-14) and a legend for symbols and components.
+
+**Figure 2/G.142 – Transmission paths at digital exchanges which include digital signal processing**
+
+### 4.2 Transmission loss through a digital exchange
+
+The 4-wire digital switching function at a digital exchange should introduce a nominal transmission loss of 0 dB. Thus, in Figure 1 (Case 1) if a 0 dBm0 sinusoidal test signal is introduced at the analogue terminals of an ideal coder connected to the input of a digital switch, a Digital Reference Sequence (DRS) should be transmitted unaltered through the switch and produce a 0 dBm0 sinusoidal signal at the analogue terminals of a decoder connected to the output of the digital switch.
+
+Except for the transmission loss considered above (and perhaps the possible loss due to exchange wiring on analogue signals), all transmission losses which are to be introduced by a digital exchange, either in a digital or analogue form, are to be governed by the applicable transmission plan (see 4.4 below).
+
+### **4.3 Relative levels**
+
+On digital paths within an all-digital network, relative levels have no real meaning or use. However, as long as a substantial portion of the worldwide telephone network is of an analogue nature, it is necessary and useful to assign relative levels to digital exchanges.
+
+The relative levels assigned to a digital exchange are applicable at the virtual international connecting points of the exchange. The virtual international connecting points are theoretical points, as explained in 3.10.1.3/G.101.
+
+In accordance with 3.10.1.4/G.101:
+
+- the send and receive relative levels at an international digital exchange should be 0 dBr, when digital facilities are being terminated;
+- the send level is 0 dBr and the receiving level is  $-0.5$  dBr at an international digital exchange, when the international exchange terminates national analogue extension facility.
+
+In the case of digital exchanges in national extensions, the send relative levels should be governed by the applicable national transmission plan.
+
+### **4.4 Echo and stability control**
+
+The overall echo and stability losses presented by a national extension are a function of the relevant transmission losses and, in the case of the use of 2-wire conversion circuits, the balance return loss introduced by the 2-wire/4-wire conversion circuit. Both contributions need to be considered in the design of digital local exchanges where there is generally scope for improving the echo and stability losses. Such improvements are likely to be needed as connections in digital networks will tend to have lower losses and longer delays than analogue connections with a consequent worsening in echo performance. As integration of ATM switches may introduce significant incremental delay, a set of guidelines for introduction of ATM technology into the PSTN has been developed and published as Recommendation G.176.
+
+### **4.4.1 Transmission loss contribution**
+
+The requirements for controlling stability and echo on international connections under all-digital or mixed analogue/digital network conditions are dealt with in Recommendation G.122. In accordance with the latter Recommendation, the national extensions are to be mainly responsible for effecting this control. Arrangements for doing so are dealt with in clause 6/G.121.
+
+Clause 6/G.121 provides the framework within which individual national transmission plans are to provide for the necessary features to effect the required control. In the case of a digital 4-wire national extension (i.e. all-digital down to the local exchange but with 2-wire analogue subscriber lines), the control can be effected entirely at the local exchange. Where the national extension is to be of a mixed analogue/digital nature, the control under some national transmission plans might be distributed among the different parts of the national extension but the main burden would in general still lie with the local exchange. Figure 1 contains examples of some of the different arrangements that might be encountered at a digital exchange.
+
+The arrangement in (Case 1) of Figure 1 deals with the termination of a digital circuit at what might be a national or an international digital exchange. In this particular case, the circuit is to be operated without introducing additional loss at the exchange.
+
+The arrangement in (Case 2) of Figure 1 also deals with the termination of a digital circuit at a national digital exchange. However, in this case, the relevant transmission plan requires that loss should be associated with the circuit at the exchange through the medium of digital pads. See 4.7 below regarding the use of digital pads.
+
+The arrangement in (Case 3) of Figure 1 deals with the termination of a 2-wire subscriber's line at a digital local exchange. The pads designated R and T are pad symbols intended to represent loss or level adjustment made in the analogue portion. Clause 6/G.121 is concerned with the appropriate choice of values for R and T.
+
+The arrangement in (Case 4) of Figure 1 is similar to that of (Case 3) except that the losses R and T are shown as being provided in the digital portion. See 4.7 below regarding the use of digital pads.
+
+The arrangement in (Cases 5, 6 and 7) of Figure 1 deals with the termination of national analogue circuits at a national or international digital exchange. In (Case 5), an analogue pad (L) is used to develop the required loss of the circuit in accordance with the relevant transmission plan. (Case 6) is similar to (Case 5) except that a digital pad (L) is used to develop the required circuit loss. (Case 7) is also similar to (Case 5) except that the analogue pad (L) as well as the A/D coder and D/A decoder are provided as part of the transmission equipment associated with the circuit rather than by equipment that is built-in as part of the switching system. Although not shown in Figure 1, the A/D coders, the D/A decoders, the 2-wire/4-wire terminating units and the pads involved in (Cases 2, 3 and 4) can also be provided as part of the transmission equipment on the transmission side of the exchange rather than by equipment that is built-in as part of the switching system.
+
+### **4.4.2 Balance return loss contribution**
+
+The contribution of balance return loss to the overall echo and stability losses is illustrated in (Cases 3 and 4) of Figure 1 which show the situation of 2-wire local lines terminating on a digital local exchange. The achieved balance return loss is determined by the match between the impedance presented by the 2-wire local line and customer terminating apparatus and the balancing impedance chosen for the digital exchange line card.
+
+In many designs of digital local exchange, there is no 2-wire switch and the 2-wire line is permanently connected to the line card. This arrangement has significant advantages for balance return loss as there is likely to be a significant reduction in the range of impedances presented to any single line card. It is then possible to choose a line card balancing impedance more closely matched to the local line impedances and obtain an improvement in balance return loss compared with the conventional compromise impedances.
+
+The optimum balancing impedance will not be the same for all Administrations as it needs to take into account the local cable types used together with the range of customer apparatus impedances. It is possible that the use of different exchange balancing impedances for different local line classes will give an improvement in performance at the expense of some increase in network administration. In general it has been found that the use of balancing networks which resemble the impedance presented by local cable give the optimum performance. Examples of balancing impedances adopted by a number of Administrations are given in Recommendation Q.552.
+
+Further improvement in balance return loss is possible where the impedance of the customer apparatus can be influenced by the Administration. Telephone instruments with an input impedance close to the impedance of the local cable can result in an improvement in the balance return loss at the digital local exchange in the order of 10 dB on short local lines. Supplement 31 to the G-series Recommendations provides principles of determining an impedance strategy for the local network. Recommendation G.126 provides guidance in regards to the balance requirements necessary to control listener echo.
+
+### 4.5 Local transmission
+
+On local calls between subscribers served by the same digital local exchange, the switching of 2-wire subscriber lines such as those shown in Figure 1, (Case 3), results in an equipment arrangement which takes on the appearance of a voice-frequency repeater – see Figure 3. As is well known, such an arrangement must include sufficient loss around the loop to provide for an adequate margin of stability. To provide for this loss, some 2-wire to 2-wire attenuation may be acceptable in some cases. The attenuation might be supported by the national transmission plan, as it provides adequate loudness rating distribution for local calls. However, in cases where the 2-wire to 2-wire attenuation is to be comparable to that generally prevailing at an analogue exchange, i.e. approximately 0 dB, adequate balance return losses must be provided at the 2-wire/4-wire junctions.
+
+Increasing the balance return losses, as referred to above, should also be beneficial to the control of echo and stability in national connections beyond the local exchange as well as on international connections.
+
+![Figure 3/G.142 – Configuration of digital local exchange on 2-wire to 2-wire connections. The diagram shows a 4-wire loop within an exchange, connected to two 2-wire lines via 2-wire/4-wire terminating units (H) and balance networks (B). The loop contains digital switches and A/D coders or D/A decoders. The 'Singing and echo path' is indicated within the loop. The 'Exchange transmission loss' is shown as a double-headed arrow across the top. A legend at the bottom defines the symbols: Digital switch (dashed line with X), A/D coder or D/A decoder (solid line with cross), Analogue pads for control of stability and echo (R and T symbols), 2-wire/4-wire terminating unit (H box), and Balance network (B box).](ff0952ef692c9d960ce5f6708bcc9711_img.jpg)
+
+Exchange transmission loss
+
+4-wire loop
+
+Singing and echo path
+
+T1208980-98
+
+Legend:
+
+- X--- Digital switch
+- +— A/D coder or D/A decoder
+- R T Analogue pads for control of stability and echo
+- H 2-wire/4-wire terminating unit
+- B Balance network
+
+Figure 3/G.142 – Configuration of digital local exchange on 2-wire to 2-wire connections. The diagram shows a 4-wire loop within an exchange, connected to two 2-wire lines via 2-wire/4-wire terminating units (H) and balance networks (B). The loop contains digital switches and A/D coders or D/A decoders. The 'Singing and echo path' is indicated within the loop. The 'Exchange transmission loss' is shown as a double-headed arrow across the top. A legend at the bottom defines the symbols: Digital switch (dashed line with X), A/D coder or D/A decoder (solid line with cross), Analogue pads for control of stability and echo (R and T symbols), 2-wire/4-wire terminating unit (H box), and Balance network (B box).
+
+Figure 3/G.142 – Configuration of digital local exchange on 2-wire to 2-wire connections
+
+### 4.6 Sidetone and input impedance
+
+Digital local exchanges can have a significant influence on the sidetone performance of telephone instruments, particularly those instruments on relatively short local lines. The reason for this can be seen in Figure 3 where the impedance presented by the exchange to the local line is a function of the input impedance of the line card and the characteristics of the singing and echo path within the exchange.
+
+For optimum sidetone performance on short local lines, the input impedance of the exchange line card should be close to the anti-sidetone impedance of the telephone instrument. In the case where the telephone instrument is designed to give good sidetone performance on long local lines, this anti-sidetone impedance is likely to be close to the characteristic impedance of the 2-wire local cable.
+
+This would lead to the digital local exchange also presenting an impedance close to that of the 2-wire local cable.
+
+On longer local lines, the exchange impedance will have less effect on the sidetone performance as the impedance presented to the telephone is masked by the local cable impedance.
+
+The final choice of exchange impedance needs to take into account a number of factors:
+
+- telephone set impedance and sensitivity characteristics;
+- local line network characteristics;
+- digital exchange current feeding arrangements,
+
+the objective being that the customer should not see a worsening in sidetone performance when connected to a digital exchange. Impedances chosen by a number of Administrations are given in Recommendation Q.552, and it is clear that there is a considerable difference between the impedances which reflect the differences between the national networks.
+
+## **4.7 Digital pads**
+
+The use of a digital pad to produce the required transmission loss in a digital path attracts a transmission penalty. This penalty has to come out of the allowance of "units of transmission impairment" allotted to the national and international portions of international connections – see 5.6/G.113. Additionally, since digital pads involve the use of digital recoding processes, the use of such pads in paths where bit integrity must be preserved is unattractive. This can be an important consideration where multi-purpose networks are contemplated. Consequently, if digital pads must be introduced, arrangements should be made to switch them out or to bypass them.
+
+## **4.8 Transmission delay**
+
+Transmission delays through digital exchanges, especially if they also perform a speech processing function, e.g. digital speech compression, or PDH/SDH to ATM interworking, could be significant. For example, such delays could have the effect of decreasing the length of connections on which echo control devices (viz. echo cancellers) should be applied. Transmission delays at digital local exchanges (or at digital PBXs) could in some cases also affect the impedance match between subscriber lines and the exchange (or PBX) in a way that could adversely affect subscriber sidetone. Transmission delays through digital exchanges should, therefore, be minimized. See Annex A/G.114 for details of the delay introduced by various items of digital equipment and systems. Recommendation G.176 provides guidance associated with the integration of ATM technology into the PSTN.
+
+For transmission delays that might be encountered at digital PDH/SDH exchanges, also see Recommendation Q.551.
+
+# ITU-T RECOMMENDATIONS SERIES
+
+|                 |                                                                                                                                |
+|-----------------|--------------------------------------------------------------------------------------------------------------------------------|
+| Series A        | Organization of the work of the ITU-T                                                                                          |
+| Series B        | Means of expression: definitions, symbols, classification                                                                      |
+| Series C        | General telecommunication statistics                                                                                           |
+| Series D        | General tariff principles                                                                                                      |
+| Series E        | Overall network operation, telephone service, service operation and human factors                                              |
+| Series F        | Non-telephone telecommunication services                                                                                       |
+| <b>Series G</b> | <b>Transmission systems and media, digital systems and networks</b>                                                            |
+| Series H        | Audiovisual and multimedia systems                                                                                             |
+| Series I        | Integrated services digital network                                                                                            |
+| Series J        | Transmission of television, sound programme and other multimedia signals                                                       |
+| Series K        | Protection against interference                                                                                                |
+| Series L        | Construction, installation and protection of cables and other elements of outside plant                                        |
+| Series M        | TMN and network maintenance: international transmission systems, telephone circuits, telegraphy, facsimile and leased circuits |
+| Series N        | Maintenance: international sound programme and television transmission circuits                                                |
+| Series O        | Specifications of measuring equipment                                                                                          |
+| Series P        | Telephone transmission quality, telephone installations, local line networks                                                   |
+| Series Q        | Switching and signalling                                                                                                       |
+| Series R        | Telegraph transmission                                                                                                         |
+| Series S        | Telegraph services terminal equipment                                                                                          |
+| Series T        | Terminals for telematic services                                                                                               |
+| Series U        | Telegraph switching                                                                                                            |
+| Series V        | Data communication over the telephone network                                                                                  |
+| Series X        | Data networks and open system communications                                                                                   |
+| Series Y        | Global information infrastructure                                                                                              |
+| Series Z        | Programming languages                                                                                                          |

marked/G/T-REC-G.228-198811-I_PDF-E/raw.md

@@ -0,0 +1,449 @@
+
+
+![ITU logo](2dfa6ac3edfe874f68aa0cbccaa42322_img.jpg)
+
+The logo of the International Telecommunication Union (ITU) features a globe with a lightning bolt superimposed on it, and the letters 'ITU' in a bold, sans-serif font.
+
+ITU logo
+
+INTERNATIONAL TELECOMMUNICATION UNION
+
+**ITU-T**
+
+**G.228**
+
+TELECOMMUNICATION  
+STANDARDIZATION SECTOR  
+OF ITU
+
+# **INTERNATIONAL ANALOGUE CARRIER SYSTEMS GENERAL CHARACTERISTICS COMMON TO ALL ANALOGUE CARRIER-TRANSMISSION SYSTEMS** ---
+
+## **MEASUREMENT OF CIRCUIT NOISE IN CABLE SYSTEMS USING A UNIFORM - SPECTRUM RANDOM NOISE LOADING**
+
+**ITU-T Recommendation G.228**
+
+(Extract from the *Blue Book*)
+
+---
+
+## NOTES
+
+1 ITU-T Recommendation G.228 was published in Fascicle III.2 of the *Blue Book*. This file is an extract from the *Blue Book*. While the presentation and layout of the text might be slightly different from the *Blue Book* version, the contents of the file are identical to the *Blue Book* version and copyright conditions remain unchanged (see below).
+
+2 In this Recommendation, the expression “Administration” is used for conciseness to indicate both a telecommunication administration and a recognized operating agency.
+
+## **MEASUREMENT OF CIRCUIT NOISE IN CABLE SYSTEMS USING A UNIFORM-SPECTRUM RANDOM NOISE LOADING**
+
+*(Geneva, 1964; further amended)*
+
+The CCITT,
+
+*considering that*
+
+(a) it is desirable to measure the performance of cable systems for frequency-division multiplex telephony under conditions closely approaching those of actual operation;
+
+(b) a signal with a continuous uniform spectrum (white noise) has statistical properties similar to those of a multiplex signal when the number of channels is not too small;
+
+(c) the use of a signal with a continuous uniform spectrum to measure the performance of such cable systems is already widespread;
+
+(d) it is necessary to standardize the frequencies and bandwidths of the measuring channels to be used for such measurements;
+
+(e) for reasons of international compatibility it is necessary to standardize the minimum attenuation and the bandwidth of the stop filters which may have to be used in the white-noise generator;
+
+(f) the CCITT has indicated, for the planning of telephone circuits, a mean value of signal power in the baseband of a multiplex telephone system to be taken into consideration during the busy hour (Recommendation G.223),
+
+*recommends that*
+
+**1** The performance of frequency-division multiplex cable systems should be measured by means of a signal with a continuous uniform spectrum in the frequency band used for the telephone channels.
+
+**2** The nominal power level of the uniform spectrum test signal should be in accordance with the conventional load, specified in Recommendation G.223. If applied at the point of interconnection of the system corresponding to  $T'$  of Recommendation G.213, the absolute power levels of interest are shown in column 4 of Table 1/G.228.
+
+**2.1** The sending equipment should be capable of providing, at the output of an inserted bandstop filter, a loading level at least up to +10 dB relative to the nominal power level defined above.
+
+**2.2** Within the bandwidth corresponding to the baseband of the system under test, the r.m.s. voltage of the white noise spectrum measured in a band of about 2 kHz should not vary by more than  $\pm 0.5$  dB. This degree of spectrum uniformity should be met in the level range up to +6 dB relative to the nominal power level, indicated in Table 1/G.228, column 4.
+
+**2.3** The white noise test signal should be available at the output of the sending equipment with a peak factor of about 12 dB with respect to the r.m.s. value.
+
+**3** The nominal effective cut-off frequencies (the cut-off frequencies of hypothetical filters having ideal square cut-off characteristics and transmitting the same power as the real filters) and tolerances for the bandpass filters proposed for the various bandwidths of systems to be tested, should be as specified in Table 2/G.228. To reduce the number of filters required, compromises have been made between the nominal effective cut-off frequency and the system bandwidth-limiting frequency in some cases. The tolerances ensure that consequent calibration errors do not exceed  $\pm 0.1$  dB and errors in measurement of intermodulation noise do not exceed  $\pm 0.2$  dB assuming system pre-emphasis of about 10 dB.
+
+TABLE 1/G.228
+
+| 1                            | 2                                      | 3                                     | 4                                                        |
+|------------------------------|----------------------------------------|---------------------------------------|----------------------------------------------------------|
+| Number of telephone channels | Relative power level at point T' (dBr) | Level of the conventional load (dBm0) | Nominal power level of the test signal at point T' (dBm) |
+| 60                           | -36                                    | 6.1                                   | -29.9                                                    |
+| 120                          | -36                                    | 7.3                                   | -28.7                                                    |
+| 300                          | -36                                    | 9.8                                   | -26.2                                                    |
+| 600                          | -36                                    | 12.8                                  | -23.2                                                    |
+|                              | -33                                    |                                       | -20.2                                                    |
+| 960                          | -36                                    | 14.8                                  | -21.2                                                    |
+|                              | -33                                    |                                       | -18.2                                                    |
+| 1 260                        | -33                                    | 16.0                                  | -17.0                                                    |
+| 1 800                        | -33                                    | 17.5                                  | -15.5                                                    |
+| 2700                         | -33                                    | 19.3                                  | -13.7                                                    |
+| 3 600                        | -33                                    | 20.5                                  | -12.5                                                    |
+| 10 800                       | -33                                    | 25.3                                  | -7.7                                                     |
+
+TABLE 2/G.228
+
+| System capacity (channels) | Limits of band occupied by telephone channels (kHz) | Effective cut-off frequencies of bandpass filters (kHz) |                  | Frequencies of recommended measuring channels (kHz)       |
+|----------------------------|-----------------------------------------------------|---------------------------------------------------------|------------------|-----------------------------------------------------------|
+|                            |                                                     | Highpass                                                | Lowpass          |                                                           |
+| 60                         | 60- 300                                             | $60 \pm 1$                                              | $300 \pm 2$      | 70 270                                                    |
+| 120                        | 60- 552                                             | $60 \pm 1$                                              | $552 \pm 4$      | 70 270 534                                                |
+| 300 {                      | 60- 1 300<br>64- 1 296 }                            | $60 \pm 1$                                              | $1 296 \pm 8$    | 70 270 534 1 248                                          |
+| 600 {                      | 60- 2 540<br>64- 2 660 }                            | $60 \pm 1$                                              | $2 600 \pm 20$   | 70 270 534 1 248 2 438                                    |
+| 960 {                      | 60- 4 028<br>64- 4 024 }                            | $60 \pm 1$                                              | $4 100 \pm 30$   | 70 270 534 1 248 2 438 3 886                              |
+| 900                        | 316- 4 188                                          | $316 \pm 5$                                             | $4 100 \pm 30$   | 534 1 248 2 438 3 886                                     |
+| 1 260 {                    | 60- 5 636<br>60- 5 564 }                            | $60 \pm 1$                                              | $5 600 \pm 50$   | 70 270 534 1 248 2 438 3 886 5 340                        |
+| 1 200                      | 316- 5 564                                          | $316 \pm 5$                                             | $5 600 \pm 50$   | 534 1 248 2 438 3 886 5 340                               |
+| 1 800 {                    | 312- 8 120<br>312- 8 204<br>316- 8 204 }            | $316 \pm 5$                                             | $8 160 \pm 75$   | 534 1 248 2 438 3 886 5 340<br>7 600                      |
+| 2 700 {                    | 312-12 336<br>316-12 388<br>312-12 388 }            | $316 \pm 5$                                             | $12 360 \pm 100$ | 534 1 248 2 438 3 886 5 340<br>7 600 11 700               |
+| 3 600 {                    | 316-17 004<br>312-16 900<br>596-17 668 }            | $316 \pm 5$                                             | $17 300 \pm 150$ | 534 or 770 1 248 2 438 3 886 5 340<br>7 600 11 700 16 400 |
+| 10 800 {                   | 4 332-59 684<br>4 404-59 580 }                      | $4 370 \pm 70$                                          | $59 600 \pm 600$ | 5 340 11 700<br>35 748 55 548                             |
+
+3.1 The discrimination of a lowpass filter should be at least 20 dB at a frequency more than 10% above nominal cut-off and at least 25 dB at frequencies more than 20% above nominal cut-off. The discrimination of a highpass filter should be at least 25 dB at frequencies more than 20% below nominal cut-off.
+
+3.2 To limit discrimination against measuring channels, the spread of losses introduced by any pair of highpass and lowpass filters should not exceed 0.2 dB over a range of frequencies which includes the upper and lower measuring channels.
+
+4 Values of the characteristics for the discrimination in each stop-band at the output of a sending equipment are given in Table 3/G.228. These characteristics are intended to apply over a temperature range from 10 °C to 40 °C.
+
+5 When the receiving equipment is connected directly to a sending equipment provided with bandstop filters which only just meet the requirements of § 4 above, the ratio of the noise power indicated by the receiving equipment when the bandstop filter is bypassed, to that indicated when the filter is in circuit, should be a minimum of 67 dB; this requirement applies when a conventional load is applied. The minimum effective bandwidth of the receiver should be 1.7 kHz; the maximum reading of absolute noise power arising from leakage given by a receiver of 1.74 kHz effective bandwidth and which just meets the foregoing leakage requirement is -85.6 dBm0p.
+
+6 Additional measuring channels may be provided by agreement between the Administrations concerned.
+
+*Note* - In Annexes A and B some general information is given on the measuring procedures, the choice of filter characteristics, correction methods and accuracy objectives.
+
+TABLE 3/G.228
+
+### **Characteristics of bandstop filters**
+
+| Centre frequency $f_0$ (kHz) | Bandwidth (kHz), in relation to $f_0$ ,<br>over which the discrimination should be at least: |        |        |        | Bandwidth (kHz)<br>in relation to $f_0$ , outside<br>of which the discrimination<br>should not exceed: |        |
+|------------------------------|----------------------------------------------------------------------------------------------|--------|--------|--------|--------------------------------------------------------------------------------------------------------|--------|
+|                              | 70 dB                                                                                        | 55 dB  | 30 dB  | 3 dB   | 3 dB                                                                                                   | 0.5 dB |
+| 70                           | ± 1.5                                                                                        | ± 2.2  | ± 3.5  | —      | ± 12                                                                                                   | ± 18   |
+| 270                          | ± 1.5                                                                                        | ± 2.3  | ± 2.9  | —      | ± 8                                                                                                    | ± 24   |
+| 534                          | ± 1.5                                                                                        | ± 3.5  | ± 7.0  | —      | ± 15                                                                                                   | ± 48   |
+| 770                          | ± 1.5                                                                                        | ± 3.8  | ± 8.0  | —      | ± 21                                                                                                   | ± 70   |
+| 1 248                        | ± 1.5                                                                                        | ± 4.0  | ± 11.0 | —      | ± 35                                                                                                   | ± 110  |
+| 2 438                        | ± 1.5                                                                                        | ± 4.5  | ± 19.0 | —      | ± 60                                                                                                   | ± 220  |
+|                              |                                                                                              | ± 15.0 | ± 30.0 | —      | ± 110                                                                                                  | ± 350  |
+| 3 886                        | ± 1.5                                                                                        |        |        |        |                                                                                                        |        |
+|                              |                                                                                              | ± 1.8  | ± 3.5  | ± 8.0  | ± 12                                                                                                   | ± 100  |
+| 5 340                        | ± 1.5                                                                                        | ± 2.2  | ± 4.0  | ± 8.5  | ± 14                                                                                                   | ± 150  |
+| 7 600                        | ± 1.5                                                                                        | ± 2.4  | ± 4.6  | ± 9.5  | ± 16                                                                                                   | ± 200  |
+| 11 700                       | ± 1.5                                                                                        | ± 3.0  | ± 7.0  | ± 11.0 | ± 20                                                                                                   | ± 300  |
+| 16 400                       | ± 1.5                                                                                        | ± 4.0  | ± 9.0  | ± 14   | ± 30                                                                                                   | ± 500  |
+| 35 748                       | ± 1.8                                                                                        | ± 2.2  | ± 3.5  | ± 5.0  | ± 20                                                                                                   | ± 150  |
+| 55 548                       | ± 2.5                                                                                        | ± 3.5  | ± 6.0  | ± 9.5  | ± 30                                                                                                   | ± 200  |
+
+*Note 1* - The characteristics recommended for the filters 70 kHz to 2438 kHz inclusive are based on coil-capacitor type filters. Those characteristics recommended for the filters at 5340 kHz and above are based on crystal-type filters. Optional characteristics are recommended for the 3886-kHz filter to permit a choice of design between a coil-capacitor type or crystal-type filter. The crystal-type filters at 35 748 kHz and 55 548 kHz are assumed to be operating in a higher harmonic mode of the crystal resonators. That is why the relative bandwidths of these filters show a discontinuity compared with the crystal-type filters up to 11 700 kHz.
+
+*Note 2* - The discrimination values quoted are relative referred to the minimum attenuation of the bandstop filters within the baseband frequency range defined by highpass and lowpass filters in Table 2/G.228. This implies that a bandstop filter suitable for measurements on one system is not necessarily suitable for measurements on a system of larger bandwidth.
+
+*Note 3* - The design of the receiver selectivity of 3886 kHz should be related to the characteristic of the crystal-type bandstop filter.
+
+*Note 4* - Due to spurious resonances, narrow spikes of the discrimination may occur in the upper passband of crystal-type bandstop filters. When resonators are operated in a higher harmonic mode narrow spikes can also appear in the lower passband. Those spikes of about 10-dB peak attenuation within 1 to 5 kHz bandwidth are admissible because they do not affect the measuring accuracy.
+
+## ANNEX A
+
+(to Recommendation G.228)
+
+## Outline of the white noise measuring method
+
+### A.1 General principle
+
+The principal components of the measuring setup are shown in Figure A-1/G.228.
+
+![Block diagram of the white noise measuring setup. It shows a sequence of blocks: 1 (Noise Generator G), 2 (Bandpass Filter B), 3 (Bandstop Filter with center frequency f_m), 4 (System under test), 5 (Receiving Bandpass Filter with bandwidth delta f and center frequency f_m), 6 (Variable Receiver Attenuator delta a), and 7 (r.m.s. level meter W). Test points T' and T are indicated before and after the system under test. Power levels P_s, W_A, and W_B are marked at specific points in the chain.](6df5629bc2fc6d82f1a1edf9d7340113_img.jpg)
+
+```
+
+graph LR
+    1[1 G] --> 2[2 B]
+    2 --> 3[3 f_m]
+    3 -- "T', n1, Ps" --> 4[4 SYSTEM]
+    4 -- "T, n2" --> 5[5 delta f, f_m]
+    5 -- "WA, WB, n2" --> 6[6 delta a]
+    6 --> 7[7 W]
+
+```
+
+Block diagram of the white noise measuring setup. It shows a sequence of blocks: 1 (Noise Generator G), 2 (Bandpass Filter B), 3 (Bandstop Filter with center frequency f\_m), 4 (System under test), 5 (Receiving Bandpass Filter with bandwidth delta f and center frequency f\_m), 6 (Variable Receiver Attenuator delta a), and 7 (r.m.s. level meter W). Test points T' and T are indicated before and after the system under test. Power levels P\_s, W\_A, and W\_B are marked at specific points in the chain.
+
+- 1 = noise generator delivering variable test signal power  $P_s$
+  - 2 = bandpass filter, effective bandwidth  $B$
+  - 3 = bandstop filter, centre frequency  $f_m$
+  - 4 = system under test
+  - 5 = receiving bandpass filter, centre frequency  $f_m$ , bandwidth  $\Delta f$  about 2 kHz
+  - 6 = variable receiver attenuator,  $\Delta a = 10 \log \frac{W_A}{W_B}$  dB
+  - 7 = r.m.s. level meter
+- $W_A$  = noise power when bandstop filter is bypassed  
+ $W_B$  = noise power when bandstop filter is inserted  
+ $\left. \begin{matrix} n_1 = \text{relative level at system input } T' \\ n_2 = \text{relative level at system output } T \end{matrix} \right\}$  (see Recommendation G.213)
+
+FIGURE A-1/G.228
+
+#### Principle of the measuring set-up
+
+### A.2 Measuring procedures
+
+Two methods for assessing the noise performance of a transmission system are in widespread use:
+
+#### A.2.1 Measurement of noise power ratio (NPR)
+
+The noise power ratio
+
+$$\text{NPR} = 10 \log \frac{W_A}{W_B} \text{ dB} = \Delta a \quad (\text{A-1})$$
+
+is measured at various levels of  $P_S$ . The r.m.s. level meter serves as an indicator only. The value  $W_A$  is the noise power in the measuring channel without taking account of the effect of frequency gaps between groups of channels in actual operation.
+
+In an  $N$ -channel system the following definitions are introduced:
+
+$$P_S = N \cdot P_{CH}$$
+
+$P_{CH}$  = variable signal power per channel
+
+$$P_{CH} = -15 \text{ dBm}0 + \Delta p = \text{load level per channel}$$
+
+$-15 \text{ dBm}0$  is the conventional load per channel according to Recommendation G.223 for systems with  $N \geq 240$ .  $\Delta p$  (dB) is the excess load relative to  $-15 \text{ dBm}0$
+
+$p_n$  = weighted noise power level (dBm0p) measured at point  $T$  in a 3.1 kHz telephone channel.
+
+The measured NPR values are usually plotted, as shown in Figure A-2/G.228, as a function of the excess channel loading  $\Delta p$ .
+
+![Figure A-2/G.228: NPR curve versus channel loading. The graph plots Noise power ratio (NPR) in dB on the y-axis (42, 46, 50) against Loading per channel (P_CH) in dBm0 on the top x-axis (-23, -19, -15, -11, -7) and Excess loading (Δp) in dB on the bottom x-axis (-8, -4, 0, +4, +8). A curve rises from approximately 42 dB at -23 dBm0 to a peak of about 49 dB at -15 dBm0, then falls to about 42 dB at -7 dBm0. A vertical line at -15 dBm0 is labeled 'Conventional load'.](ab9eb65dcbc2696ec77b66f70f9e0e91_img.jpg)
+
+| Excess loading ( $\Delta p$ ) [dB] | Loading per channel ( $P_{CH}$ ) [dBm0] | Noise power ratio (NPR) [dB] |
+|------------------------------------|-----------------------------------------|------------------------------|
+| -8                                 | -23                                     | 42                           |
+| -4                                 | -19                                     | 46                           |
+| 0                                  | -15                                     | 49                           |
+| +4                                 | -11                                     | 46                           |
+| +8                                 | -7                                      | 42                           |
+
+Figure A-2/G.228: NPR curve versus channel loading. The graph plots Noise power ratio (NPR) in dB on the y-axis (42, 46, 50) against Loading per channel (P\_CH) in dBm0 on the top x-axis (-23, -19, -15, -11, -7) and Excess loading (Δp) in dB on the bottom x-axis (-8, -4, 0, +4, +8). A curve rises from approximately 42 dB at -23 dBm0 to a peak of about 49 dB at -15 dBm0, then falls to about 42 dB at -7 dBm0. A vertical line at -15 dBm0 is labeled 'Conventional load'.
+
+FIGURE A-2/G.228  
+NPR curve versus channel loading
+
+The relation between NPR values measured on a channel and the weighted noise power level referred to a zero relative level point is:
+
+$$P_n = (- \text{NPR} - 18.6 - 10 \log k + \Delta p) \text{ dBm0p} \quad (\text{A-2})$$
+
+$k = B/4N$  ( $B$  in kHz) is a correction factor which takes account of the effect of the frequency gaps between groups of channels in the transmission system.
+
+Table A-1/G.228 gives examples of the correction for some N-channel systems:
+
+TABLE A-1/G.228
+
+|                  |      |      |      |        |
+|------------------|------|------|------|--------|
+| $N$              | 300  | 960  | 2700 | 10 800 |
+| $10 \log k$ (dB) | 0.14 | 0.22 | 0.46 | 1.08   |
+
+#### A.2.2 *Direct measurement of weighted noise power level*
+
+With the particular choice of the effective receiver bandwidth
+
+$$\Delta f = 1.74 \text{ kHz} (= 3.1 \text{ kHz} \cdot 10^{-0.25}),$$
+
+the weighted noise power  $P_n$  in a telephone channel is:
+
+$$P_n = W_B \text{ (see Figure A-1/G.228)}$$
+
+and the weighted noise level  $p_n$ , referred to a point of zero relative level becomes:
+
+$$p_n = \left[ 10 \log \frac{W_B}{1 \text{ mW}} + n_2 \text{ (dBr)} \right] \text{ dBm0p} \quad (\text{A-3})$$
+
+In this case the receiver (component 7 of Figure A-1/G.228) must be a calibrated power level meter.
+
+### A.3 *Examples of investigations using the white noise measuring method*
+
+Two kinds of investigations can be made on a system (with length  $L$ ) between flat relative level points  $T'$  and  $T$ . The one [case a)] investigates the effect on the noise performance of load deviations at the input of the system, whereas the other [case b)] indicates the influence of level misalignments along the transmission line:
+
+- The test signal noise power  $P_S$  is varied and the weighted noise level  $p_n$  is determined in dBm0p. The result is plotted as indicated in Figure A-3/G.228.
+
+Alternatively to the indication of the noise level for system length  $L$  in dBm0p, the noise power could have been indicated in pW0p/km.
+
+![Graph showing Weighted noise level (p_n) in dBm0p versus Loading per channel (P_CH) in dBm0 and Excess loading (Δp) in dB. The curve shows a sharp increase in noise level as loading increases beyond the conventional load of -15 dBm0.](352c5fab6f936356e9570761a02ab71e_img.jpg)
+
+The graph plots Weighted noise level ( $p_n$ ) in dBm0p on the y-axis (ranging from -68 to -52) against Loading per channel ( $P_{CH}$ ) in dBm0 on the top x-axis (ranging from -23 to -3) and Excess loading ( $\Delta p$ ) in dB on the bottom x-axis (ranging from -8 to +16). A curve starts at approximately -68 dBm0p at -23 dBm0 loading and rises sharply, passing through -60 dBm0p at -15 dBm0 loading (labeled 'Conventional load') and reaching -52 dBm0p at -7 dBm0 loading. The curve continues to rise steeply towards -3 dBm0 loading. The bottom x-axis is labeled 'Excess loading ( $\Delta p$ )' and 'CCITT-45170'.
+
+Graph showing Weighted noise level (p\_n) in dBm0p versus Loading per channel (P\_CH) in dBm0 and Excess loading (Δp) in dB. The curve shows a sharp increase in noise level as loading increases beyond the conventional load of -15 dBm0.
+
+FIGURE A-3/G.228
+
+**Weighted noise power level versus system load (relative levels of the system fixed according to plan)**
+
+- b) The relative levels on the transmission line are varied by insertion of attenuators  $-\Delta n$  and  $+\Delta n$  at the input and output of the system as is illustrated in Figure A-4/G.228 which is an excerpt of Figure A-1/G.228.
+
+![Block diagram showing a transmission system with input and output attenuators. The input attenuator is labeled -Δn and the output attenuator is labeled +Δn. The system is labeled SYSTEM. The input level is n1 and the output level is n2. The levels at the input and output of the system are n1 + Δn and n2 + Δn respectively.](7a0db9703b68b3d06cdaeefc084c0006_img.jpg)
+
+The diagram shows a transmission system block labeled 'SYSTEM'. At the input (point  $T'$ ), there is an attenuator labeled  $-\Delta n$ . The input level is  $n_1$ . The level at the input of the system is  $n_1 + \Delta n$ . At the output of the system, there is an attenuator labeled  $+\Delta n$ . The level at the output of the system is  $n_2 + \Delta n$ . The output level is  $n_2$ . The diagram is labeled 'CCITT-45180'.
+
+Block diagram showing a transmission system with input and output attenuators. The input attenuator is labeled -Δn and the output attenuator is labeled +Δn. The system is labeled SYSTEM. The input level is n1 and the output level is n2. The levels at the input and output of the system are n1 + Δn and n2 + Δn respectively.
+
+FIGURE A-4/G.228
+
+**Variation of the internal relative levels of the system**
+
+The test signal noise power  $P_S$  is set to the conventional value (-15 dBm0/4 kHz) at point  $T'$  and is kept constant. The noise power level in the measuring channel is determined at point  $T$  as a function of the relative level at the repeater output, for example. The result is plotted as shown in Figure A-5/G.228.
+
+![FIGURE A-5/G.228: Weighted noise level in measuring channel versus relative level at repeater output. The graph shows a U-shaped curve representing the weighted noise level (p_n) in dBm0p against the relative level at the repeater output in dBr and dB. The x-axis has three scales: dBr from -26 to -10, Relative level at the repeater output from -8 to +8, and Deviation of relative level from nominal value from -8 to +8 dB. The y-axis shows Weighted noise level (p_n) in dBm0p from -68 to -60. A vertical line at -18 dBr indicates the 'Nominal relative level'. The curve reaches its minimum of approximately -68 dBm0p at 0 dB deviation from the nominal value.](adb1f42239329fa8283d1a40005f989f_img.jpg)
+
+| Relative level (dBr) | Deviation from nominal (dB) | Weighted noise level (dBm0p) |
+|----------------------|-----------------------------|------------------------------|
+| -26                  | -8                          | -60.5                        |
+| -22                  | -4                          | -65.0                        |
+| -18                  | 0                           | -68.0                        |
+| -14                  | +4                          | -65.5                        |
+| -10                  | +8                          | -60.0                        |
+
+FIGURE A-5/G.228: Weighted noise level in measuring channel versus relative level at repeater output. The graph shows a U-shaped curve representing the weighted noise level (p\_n) in dBm0p against the relative level at the repeater output in dBr and dB. The x-axis has three scales: dBr from -26 to -10, Relative level at the repeater output from -8 to +8, and Deviation of relative level from nominal value from -8 to +8 dB. The y-axis shows Weighted noise level (p\_n) in dBm0p from -68 to -60. A vertical line at -18 dBr indicates the 'Nominal relative level'. The curve reaches its minimum of approximately -68 dBm0p at 0 dB deviation from the nominal value.
+
+FIGURE A-5/G.228
+
+**Weighted noise level in measuring channel versus relative level at repeater output**
+
+# ANNEX B
+
+(to Recommendation G.228)
+
+### **Measuring accuracy considerations affecting the design of the measuring equipment**
+
+### **B.1 Introduction**
+
+The Recommendations relating to the measurement of circuit noise in systems artificially loaded with uniform spectrum random noise simulating FDM telephone signals were agreed after carefully coordinated studies by three CCI Study Groups concerned. The different Recommendations provided for the application of the white noise measuring method to cable systems (CCITT Recommendation G.228), radio-relay systems (CCIR Recommendation 399 [1]), satellite systems (CCIR Recommendation 482 [2]) and translating equipments (CCITT Recommendation G.230). The objective of the coordination was that the separately recommended measuring equipments should conform with common measuring accuracy objectives and, as far as possible, be compatible and interchangeable.
+
+The overall accuracy objective of the measuring equipment when used for routine maintenance measurements is  $\pm 2$  dB. A higher accuracy of about  $\pm 1$  dB is desirable when measurements are made for the purpose of assessing the noise performance of a system in relation to required performance. This can be achieved by following certain procedures and applying corrections as described in B.4 and B.5 below.
+
+This Annex states how certain characteristics of measuring equipments were related to measuring accuracy objectives; any future extensions of the Recommendations to provide for measurements on new transmission systems, as yet unstandardized, should take account of those relationships.
+
+### **B.2 Bandstop fillers**
+
+#### **B.2.1 Choice of centre frequencies**
+
+In all cases the choice of nominal centre frequencies of band-elimination filters (i.e. of measuring channels) should take account of the need to minimize the combined discrimination of the pair of bandpass filters used when the
+
+bandstop filter provides a lower or upper measuring channel. Therefore, as a rule the centre frequency of a lower measuring channel should be at least 15% above the effective cut-off frequency of the highpass filter and the centre frequency of an upper measuring channel should be more than approximately 5% below the cut-off frequency of the lowpass filter involved. Under § 3.2 of the text of this Recommendation it is prescribed that "the spread of losses introduced by any pair of highpass and lowpass filters should not exceed 0.2 dB over a range of frequencies which includes the outer measuring channels".
+
+#### B.2.2 *Leakage*
+
+The discrimination of a bandstop filter in the neighbourhood of the centre frequency determines, jointly with the receiver selectivity the smallest noise-to-signal ratio that can be measured accurately, i.e. the "leakage" effect. The bandstop filter discrimination of 70 dB (Table 3/G.228) results in a ratio of the order of -67 dB-being measured when the noise is actually negligible. Leakage effect in the receiver is adequately limited by requiring (see § 5 in the text of the Recommendation) that the NPR value should be a minimum of 67 dB when connected directly to a send equipment with bandstop filters which only just meet the discrimination requirements of Table 3/G.228 and when a conventional load of -15 dBm/4 kHz is applied.
+
+*Note* - According to Formula (A-2) of Annex A this value of NPR = 67 dB corresponds to a residual noise level of -85.6 dBm0p (i.e. 2.8 pW0p) at the most.
+
+#### B.2.3 *Effective bandwidth*
+
+The basic requirement for the stopband is the condition that the discrimination should be at least 70 dB in a bandwidth of at least 3 kHz. The effective bandwidths (approximately the 3-dB points) recommended in Table 3/G.228 have been found to be technically feasible and lie in the order of 5% or less of the system bandwidth with coil-capacitor type filters and are less than 0.5% with crystal-type filters. It would present economic difficulties to reduce the relative bandwidth of the coil-type filters or to increase the relative bandwidth of the crystal-type filters.
+
+##### B.2.3.1 *Third order nonlinearity products*
+
+The attenuation of the noise loading signal in the vicinity of the measuring channel introduced by a bandstop filter causes an under-indication reading, erring on the low side, of third order nonlinearity noise power in that measuring channel. This under-indication is directly proportional to the effective bandwidth of the elimination filter.
+
+Assuming that procedures B.4.3 and B.4.4 below are both observed, the under-indication of third order products in a system using no pre-emphasis is about 0.05 dB for a top measuring channel filter, the effective bandwidth of which is 1% of the system bandwidth. The error associated with a particular filter is at its maximum when the filter provides the top measuring channel of a system. When the same filter is used in wider band systems (thus corresponding to an intermediate measuring channel of the system) its bandwidth is a smaller proportion of the system bandwidth and the associated error is smaller.
+
+When pre-emphasis is used but total signal power is unchanged the error is increased by the ratio of the signal density near the measuring channel of the pre-emphasized system to that of the system without pre-emphasis.
+
+The effective bandwidths of crystal-type bandstop filters are so small that their effect on measurement errors is negligible.
+
+The recommended effective bandwidths for coil-capacitor bandstop filters (Table 3/G.228) are such that the under-indication of third order nonlinear noise powers, when the filters provide top measuring channels of systems without pre-emphasis, falls in the range 0.25 to 0.30 dB. This range of errors becomes 0.60 to 0.90 dB for systems emphasized by 8 to 10 dB as is the case in FDM radio-relay systems (CCIR Recommendation 275 [3]) or in wideband systems on coaxial cables.
+
+##### B.2.3.2 *Second order nonlinearity products*
+
+In long transmission systems third order nonlinearity products normally form a more significant proportion of the total system noise than those of second order. For this reason the recommended maximum effective bandwidths of bandstop filters have been determined on the basis of accuracy objectives for the measurement of third order nonlinearity products.
+
+Nevertheless, measuring equipments may still be used for investigations of cases where second order nonlinearity products dominate. Corrections for known filter bandwidths may be made on the following basis:
+
+- a) Again assuming that procedures B.4.3 and B.4.4 below are observed, the error in a reading of second order nonlinearity products introduced by the bandstop filter is an excess reading, rather than the under-indication in the case of third order nonlinearity products.
+- b) The excess reading is directly proportional to the effective bandwidth of the bandstop filter expressed as a percentage of the system bandwidth. The approximate proportionality, assuming no system pre-emphasis:
+  - for measuring channels located near the lower limit of the system bandwidth, an effective bandwidth of 1% system bandwidth causes an excess reading of 0.05 dB for second order intermodulation noise power;
+  - for measuring channels located in the middle, or near the upper limit, of the system bandwidth, an effective bandwidth of 1% system bandwidth causes an excess reading of 0.1 dB.
+- c) The effect of system pre-emphasis in the case of a bandstop filter near the lower limit of the system bandwidth, i.e. where the density of second order nonlinearity products tends to be greatest, is to reduce the error attributable to a given filter bandwidth in the same proportion that the signal density at that frequency is reduced by pre-emphasis.
+
+### B.3 *Bandpass filters*
+
+In order to reduce the number of different filters, compromises have been made in some cases between the nominal effective cut-off frequency and the system bandwidth limiting frequency (cf. § 3 of the text).
+
+For the larger systems there may also be a significant difference between the frequency bandwidth  $4 N$  kHz ( $N$  being the system capacity expressed in telephone channels) and the system bandwidth (Table 2/G.228).
+
+Both these facts are taken into account by the correction factor  $k$  introduced in equation (A-2) of Annex A and in Table A-1/G.228.
+
+The recommended tolerances on the nominal values of cut-off frequencies are such that the actual and nominal bandwidths of the signal load cannot differ by more than 1%. This ensures that calibration errors (in NPR measurements) due to this particular imperfection do not exceed about 0.05 dB.
+
+The tolerances on the effective lowpass cut-off frequencies are in all cases less than 1.0% of the nominal system bandwidth and in most cases less than 0.8%. A difference of 0.8% leads to an error, in third order nonlinearity noise measurement, of 0.1 dB, this allowing for a pre-emphasis of 8 dB. Even allowing for a greater degree of pre-emphasis, the maximum error from this cause should not exceed 0.15 dB.
+
+### B.4 *Procedures for high accuracy measurements*
+
+The following measuring procedures are recommended for high accuracy type of measurements, for example checks that transmission system noise performance objectives are being achieved.
+
+#### B.4.1 *Signal load adjustment*
+
+The loading power should be adjusted to the nominal value by means of a true r.m.s. level measuring device. The maximum error, including reading error, should not exceed  $\pm 0.15$  dB.
+
+#### B.4.2 *Receiver calibration*
+
+B.4.2.1 Using the NPR method (§ A.2.1) the receiver should be set with reference to the received signal immediately before insertion of a bandpass filter.
+
+B.4.2.2 Using the direct noise power measuring method (§ A.2.2) the receiver calibration error could be decreased to  $\pm 0.15$  dB at the particular measuring slot by checking the reading with the aid of a white noise signal and a d.c.-calibrated true r.m.s. level meter.
+
+*Note* - The accuracy of measurements related to the zero relative level point (dBm0p or pW0p) also depends on how precisely the relative level at the measuring point ( $n_2$  of Figure A-1/G.228) is known.
+
+#### B.4.3 *Insertion of bandstop filters*
+
+Only one bandstop filter should be inserted at a time. This limits errors in measurement of intermodulation noise.
+
+#### B.4.4 *Readjustment of signal load*
+
+Normally, the signal load should be readjusted to the nominal value after the insertion of a bandstop filter. When measurements are specifically to investigate second-order intermodulation, or when this is known to dominate, greater accuracy is obtained by readjusting only for the specified passband insertion loss of the bandstop filter and not for the loss of spectrum energy in the measuring slot.
+
+*Note* - The effect of the measuring slot bandwidth is negligible with crystal-type bandstop filters.
+
+#### B.4.5 *Measurement at the receiver*
+
+B.4.5.1 Using the NPR method the noise power ratio is now measured as the change required in the setting of an attenuator ( $\Delta a$  in Figure A-1/G.228) to restore the pointer of the indicating instrument to the original setting.
+
+B.4.5.2 Using the direct measuring method the weighted noise level can be read in dBmp (or pWp) from the instrument. Optional means may be provided, e.g. to shift the calibration by setting a switch to the relative level  $n_2$  of the measuring access point  $T$  so that the dBm0p or pW0p values are indicated.
+
+### B.5 *Corrections for high accuracy measurements*
+
+The effects of the following error sources can be reduced by applying corrections to the measured values:
+
+#### B.5.1 *Receiver calibration in connection with NPR method*
+
+##### B.5.1.1 *Irregularity of the noise source*
+
+The tolerance for the spectrum regularity is  $\pm 0.5$  dB. A calibration table (or curve) should be available for each noise generator.
+
+##### B.5.1.2 *Errors of effective system bandwidth*
+
+A correction in the conversion of NPR values into noise levels (in dBm0p) by application of the correction factor  $k$  in equation (A-2) allows first, for the difference between nominal occupied bandwidth of the system under test and actual bandwidth  $B$  between bandpass filter effective cut-off frequencies and secondly, for the difference between nominal occupied bandwidth and the total bandwidth actually occupied by telephone channels (i.e.  $4N$  kHz).
+
+##### B.5.1.3 *Passband attenuation distortion of bandpass filters at the measuring frequency*
+
+The corrections in §§ B.5.1.1 and B.5.1.2 should ensure calibration to an accuracy of  $\pm 0.2$  dB.
+
+#### B.5.2 *Bandstop filter effects*
+
+If coil-capacitor type bandstop filters are used, it might be worthwhile to assess the error of the measured intermodulation noise due to the effective bandwidth of these filters. To this end the rules quoted in B.2.3.1 and B.2.3.2 above should be applied.
+
+Approximate corrections for this error are thus possible when the proportion of third- and second-order intermodulation noise has been determined.
+
+### B.6 *Limitations of the noise loading measurement technique*
+
+B.6.1 Very low noise levels of less than about -83 dBm0p (about 5 pW0p) cannot be expected to be measured with an error of less than 2 dB, where the inherent noise leakage of the white noise measuring set is at the limit corresponding to  $\text{NPR} \geq 67$  dB as explained in B.2.2 above.
+
+B.6.2 Although the measurements made at the specified frequencies may confirm that the design objectives are met, the noise performance of a system between these frequencies cannot always be inferred accurately from these measurements. Whether this interpolation is justified or not has to be established for the system under consideration. An approximate indication of the frequency dependency can be gained from the frequency characteristic of the basic noise (without loading) which can be measured with the aid of a selective level meter and continuously varying the frequency. The total noise performance of a system may be evaluated, when necessary, by carrying out measurements using additional test equipment.
+
+## **Bibliography on accuracy of white noise measuring methods**
+
+MUELLER (M.): Noise loading test errors due to finite slot width, *Data and Communications design*, pp. 20-24, March-April 1973.
+
+SPINDLER (W.): Noise loading measuring procedures and error sources, *Telecommunications*, pp. 32C-32F, July 1974.
+
+## **References**
+
+- [1] CCIR Recommendation *Measurement of noise using a continuous uniform spectrum signal on frequency-division multiplex telephony radio-relay systems*, Vol. IX, Rec. 399, Dubrovnik, 1986.
+- [2] CCIR Recommendation *Measurement of performance by means of a signal of a uniform spectrum for systems using frequency-division multiplex telephony in the fixed-satellite service*, Vol. IV, Rec. 482, Dubrovnik, 1986.
+- [3] CCIR Recommendation *Pre-emphasis characteristic for frequency modulation radio-relay systems for telephony using frequency-division multiplex*, Vol. IX, Rec. 275, Dubrovnik, 1986.